Designing and prototyping WebRTC and IMS integration using open source tools

WebRTC, or Web Real-time Communications, is a collection of web standards that detail the mechanisms, architectures and protocols that work together to deliver real-time multimedia services to the web browser. It represents a significant shift from the historical approach of using browser plugins, which over time, have proven cumbersome and problematic. Furthermore, it adopts various Internet standards in areas such as identity management, peer-to-peer connectivity, data exchange and media encoding, to provide a system that is truly open and interoperable. Given that WebRTC enables the delivery of multimedia content to any Internet Protocol (IP)-enabled device capable of hosting a web browser, this technology could potentially be used and deployed over millions of smartphones, tablets and personal computers worldwide. This service and device convergence remains an important goal of telecommunication network operators who seek to enable it through a converged network that is based on the IP Multimedia Subsystem (IMS). IMS is an IP-based subsystem that sits at the core of a modern telecommunication network and acts as the main routing substrate for media services and applications such as those that WebRTC realises. The combination of WebRTC and IMS represents an attractive coupling, and as such, a protracted investigation could help to answer important questions around the technical challenges that are involved in their integration, and the merits of various design alternatives that present themselves. This thesis is the result of such an investigation and culminates in the presentation of a detailed architectural model that is validated with a prototypical implementation in an open source testbed. The model is built on six requirements which emerge from an analysis of the literature, including previous interventions in IMS networks and a key technical report on design alternatives. Furthermore, this thesis argues that the client architecture requires support for web-oriented signalling, identity and call handling techniques leading to a potential for IMS networks to natively support these techniques as operator networks continue to grow and develop. The proposed model advocates the use of SIP over WebSockets for signalling and DTLS-SRTP for media to enable one-to-one communication and can be extended through additional functions resulting in a modular architecture. The model was implemented using open source tools which were assembled to create an experimental network testbed, and tests were conducted demonstrating successful cross domain communications under various conditions. The thesis has a strong focus on enabling ordinary software developers to assemble a prototypical network such as the one that was assembled and aims to enable experimentation in application use cases for integrated environments

Large-Scale Measurement of Real-Time Communication on the Web

Web Real-Time Communication (WebRTC) is getting wide adoptions across the browsers (Chrome, Firefox, Opera, etc.) and platforms (PC, Android, iOS). It enables application developers to add real-time communications features (text chat, audio/video calls) to web applications using W3C standard JavaScript APIs, and the end users can enjoy real-time multimedia communication experience from the browser without the complication of installing special applications or browser plug-ins. As WebRTC based applications are getting deployed on the Internet by thousands of companies across the globe, it is very important to understand the quality of the real-time communication services provided by these applications. Important performance metrics to be considered include: whether the communication session was properly setup, what are the network delays, packet loss rate, throughput, etc. At Callstats.io, we provide a solution to address the above concerns. By integrating an JavaScript API into WebRTC applications, Callstats.io helps application providers to measure the Quality of Experience (QoE) related metrics on the end user side. This thesis illustrates how this WebRTC performance measurement system is designed and built and we show some statistics derived from the collected data to give some insight into the performance of today’s WebRTC based real-time communication services. According to our measurement, real-time communication over the Internet are generally performing well in terms of latency and loss. The throughput are good for about 30% of the communication sessions

Analysis of RTCWeb Data Channel Transport Options

The Web has introduced a new technology in a more distributed and collaborative form of communication, where the browser and the user replace the web server as the nexus of communications in a way that after the call establishment through web servers, the communication is performed directly between browsers as peer to peer fashion without intervention of the web servers. The goal of Real Time Collaboration on the World Wide Web (RTCWeb) project is to allow browsers to natively support voice, video, and gaming in interactive peer to peer communications and real time data collaboration. Several transport protocols such as TCP, UDP, RTP, SRTP, SCTP, DCCP presently exist for communication of media and non-media data. However, a single protocol alone can not meet all the requirements of RTCWeb. Moreover, the deployment of a new transport protocol experiences problems traversing middle boxes such as Network Address Translation (NAT) box, firewall. Nevertheless, the current implementation for transportation of non-media in the very first versions of RTCWeb data does not include any congestion control on the end-points. With media (i.e., audio, video) the amount of traffic can be determined and limited by the codec and profile used during communication, whereas RTCWeb user could generate as much as non-media data to create congestion on the networks. Therefore, a suitable transport protocol stack is required that will provide congestion control, NAT traversal solution, and authentication, integrity, and privacy of user data. This master's thesis will give emphasis on the analysis of transport protocol stack for data channel in RTCWeb and selects Stream Control Transmission Protocol (SCTP), which is a reliable, message oriented general-purpose transport layer protocol, operating on top of both IPv4 and IPv6, providing congestion control similar to TCP and additionally, some new functionalities regarding security, multihoming, multistreaming, mobility, and partial reliability. However, due to the lack of universal availability of SCTP within the OS(s), it has been decided to use the SCTP userland implementation. WebKit is an open source web browser engine for rendering web pages used by Safari, Dashboard, Mail, and many other OS X applications. In WebKit RTCWeb implementation using GStreamer multimedia framework, RTP/UDP is utilized for the communication of media data and UDP tunnelling for non-media data. Therefore, in order to allow a smooth integration of the implementation within WebKit, we have decided to implement GStreamer plugins using SCTP userland stack. This thesis work also investigates the way Mozilla has integrated those protocols in the browser's network stack and how the Data Channel has been designed and implemented using SCTP userland stack

Techno-Economic Feasibility of Web Real-Time Communications

WebRTC is an ongoing effort to build an open framework for real-time audio and video communication capabilities that turn Web browsers, and other clients supporting it, into a platform for person-to-person communication. Previously, real-time communication (RTC) has been achievable in the Web browser only by installing third party software. WebRTC brings native support for RTC to the Web browsers and exposes it freely to web developers via standardized JavaScript API. This brings RTC as a feature to the Web, which can foster further innovation. This thesis studies the techno-economic feasibility of WebRTC with the help of a framework for feasibility analysis of Internet protocols, developed by Levaä and Suomi (2013). To provide input for the framework, we conduct an interview study, as well as research of available Web resources. Further, we explore what market opportunities may arise, provided that WebRTC is successfully adopted. To do that, we use Value Network Configurations as a tool for studying and visualizing the possible relationships between market players and the roles they assume in the ecosystem. We find that WebRTC is a feasible technology in its basic, but highly relevant use case of one-to-one browser-to-browser communication. While we discover a number of unresolved challenges, we do not see any insurmountable obstacles that would prevent WebRTC adoption. WebRTC opens up opportunities for companies that would use it directly to deliver an RTC service, but also creates space for WebRTC PaaS providers in the market. Additionally, WebRTC interconnecting with legacy systems, such as PSTN or PLMN, opens up opportunity for telecom operators to explore creating new ways of communication for their customers

Desenvolvimento de uma aplicação colaborativa baseada em WebRTC

Mestrado em Sistemas de InformaçãoA comunicação desenrolou um papel fundamental na evolução do ser humano. Com o advento dos telefones tornou-se possível comunicar à distância, mas apenas a voz era transmitida. O desenvolvimento das tecnologias permitiu posteriormente a troca de vídeo entre dois pontos longínquos, mas as infra-estruturas eram limitadas. A Internet veio oferecer a permuta de informação de forma eficiente e adaptável, características apelativas para as comunicações em tempo real. A banalização deste conjunto de tecnologias permitiu às empresas baixar os seus custos ao integrar a telefonia com esse mesmo conjunto. Esta acção tornou-se uma necessidade proveniente da crise económica instalada nos últimos anos. Nesta mudança acrescenta-se o benefício das entidades empresariais poderem desenvolver interacções intrínsecas entre os seus serviços e a telefonia. Os aperfeiçoamentos aos conteúdos multimédia continuam actualmente a vários níveis, sejam equipamentos ou mecanismos dedicados à qualidade dos mesmos, tudo devido às implicações das comunicações em tempo-real. Uma parte interessante deste progresso é o uso da voz e vídeo em diversos ambientes colaborativos, como reuniões corporativas, jogos online ou actividades lúdicas. Para estes fins, a diversidade de aplicações é crescente mas ainda limitada, requerendo conhecimentos de instalação ou configuração que podem criar dificuldades de usabilidade ao utilizador típico da Internet. Neste documento é proposta uma solução capaz de minimizar os obstáculos que as soluções actuais apresentam aos seus utilizadores. Baseada em HTML5, esta aplicação oferece um serviço onde três ou mais intervenientes têm a habilidade de comunicar e colaborar entre si, com recurso exclusivo ao seu browser. Será realizado um estudo das tecnologias web emergentes para adquirir as bases tecnológicas essenciais a serem implementadas no sistema designado.Communication unrolled a key role in human evolution. With the advent of mobile communications it became possible to communicate at a distance, but only the voice was transmitted. Later technology development allowed the exchange of video between two distant points, but the infrastructure was limited. The Internet has to offer exchange information efficiently and adaptively, appealing features for real-time communications. The banality of this set of technologies enabled companies to lower their costs by integrating telephony for the same. This action has become a necessity installed from the economic crisis in recent years. This change builds up the benefit of the business entities that can conceive close interactions between its services and the media referred. The improvements to multimedia content currently continue at various levels, equipment or mechanisms are dedicated to the quality of them, all due to the implications of communications in real-time. An interesting part of this progress is the application of voice and video in multiple collaborative environments, such as business meetings, online games or play activities. For these purposes, the range of applications is growing but still limited, requiring knowledge of installation or configuration, creating difficulties to the typical Internet user. In this document it’s proposed a solution that would minimize the obstacles that current solutions present to its users. Based on HTML5, this application offers a service where three or more participants have the ability to communicate and collaborate requiring only their browser. A detailed study of emerging web technologies will be made to acquire the essential technological bases to be implemented on the target system

Browser to browser media streaming with HTML5

Video on demand services generate one of the largest portions of Internet traffic every day and their use is constantly increasing. Scaling up the infrastructure to meet this demand with the current model of Internet video delivery over HTTP, is proving to be very costly for service providers. An alternative model for video content delivery is the need of the hour to meet this challenge. Peer-to-peer streaming is a viable alternative model that is highly scalable and can meet this increasing demand. The emerging HTML5 standard introduces APIs that give Web browsers the ability to communicate directly with each other in real-time. This also allows web browsers to behave as Peer-to-peer nodes. In this thesis, we utilize these new APIs to develop a Video on demand service within the Web browser. The goal of this being to determine the feasibility of such a solution and evaluate the usage of these APIs. We hope to aid the HTML standardization process with our findings

Multimedia congestion control: circuit breakers for unicast RTP sessions

The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows. This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms

Performance analysis of topologies for Web-based Real-Time Communication (WebRTC)

Real-time Communications over the Web (WebRTC) is being developed to be the next big improvement for rich web applications. This enabler allow developers to implement real-time data transfer between browsers by using high level Application Programing Interfaces (APIs). Running real-time applications in browsers may lead to a totally new scenario regarding usability and performance. Congestion control mechanisms may influence the way this data is sent and metrics such as delay, bit rate and loss are now crucial for browsers. Some mechanisms that have been used in other technologies are implemented in those browsers to handle the internals of WebRTC adding complexity to the system but hiding it from the application developer. This new scenario requires a deep study regarding the ability of browsers to adapt to those requirements and to fulfill all the features that are enabled. We investigate how WebRTC performs in a real environment running over an current web application. The capacity of the internal mechanisms to adapt to the variable conditions of the path, consumption resources and rate. Taking those principles, we test a range of topologies and use cases that can be implemented with the current version of WebRTC. Considering this scenario we divide the metrics in two categories, host and network indicators. We compare the results of those tests with the expected output based on the defined protocol in order to evaluate the ability to perform real-time media communication over the browser

Study, design and implementation of WebRTC for a realtime multimedia messaging application

Social networks are no longer a phenomenon; nowadays it is not that they are a reality but have become something indispensable. During its growth and consolidations period internet has suffered a great transformation due to the new kind of most demanded content. Sharing images, videos or even making calls with another user are tasks that an average user would make several times a day. This transition could only happen thanks to new technologies that not only simplify those tasks but, due to handheld devices' irruption, would work successfully under reasonable data and battery consumption rates. Videoconferences over the network and multimedia data streams in general have always gone hand in hand of closed software products like Macromedia Flash, for instance, that required of a plugin installation on the browser by the end user. Under those premises, this project will focus on the investigation of WebRTC as a technology capable of successfully achieving videoconferences between users without the need of any browser plugin. In order to verify the knowledge gathered through the study of the technology, the design, architecture and implementation of an application capable of doing so will be proposed

How far are we from WebRTC-1.0? An update on standards and a look at what's next

Real-time communication between browsers has represented an unprecedented standardization effort involving both the IETF and the W3C. These activities have involved both the real-time protocol suite and the application-level JavaScript APIs to be offered to developers in order to allow them to easily implement interoperable real-time multimedia applications in the web. This article sheds light on the current status of standardization, with special focus on the upcoming final release of the so-called WebRTC-1.0 standard ecosystem. It takes stock of the situation with respect to hot topics such as codecs, session description and stream multiplexing. It also briefly discusses how standard bodies are dealing with seamless integration of the initially competing effort known as “Object Real Time Communications.