A Flexible and Scalable Media Centric Webrtc
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Vidyo® Server for Webrtc Click to Collaborate
Datasheet Vidyo® Server for WebRTC Click to Collaborate The Vidyo Server for WebRTC extends the Vidyo platform to include WebRTC capable browsers. Now join Vidyo conferences right from their web browser without any software installation. With a simple click-to-connect link, participants can enjoy up to HD video quality. The Vidyo Server for WebRTC is fully integrated with the Vidyo platform. This means participants joining through WebRTC can enjoy the interoperability delivered by the Vidyo platform including native Vidyo endpoints as well as third party H.323, SIP, and Microsoft® Skype for Business®. Deployed as a virtual machine, the Vidyo Server for WebRTC can be easily managed and scaled to meet demand. Concurrency is determined by flexible VidyoLine licenses that can float between native Vidyo endpoints and WebRTC clients as needed. Calls are secured through encryption using HTTPS, DTLS, and SRTP. Key Features Incredible User Experience Easy to Deploy & Manage • Native WebRTC support for plug-in free • Virtual server for easy deployment in your videoconferencing data center, colocation facility or in the cloud • Support for non-WebRTC browsers via • Dynamically scalable capacity based on VidyoWeb plug-in provisioned resources • Full two-way video communications on • Spin up new instances of Vidyo Server for ChromeBooks WebRTC to rapidly cluster and add capacity • Multipoint video layouts with up to 6 viewable • Simplify administration, configuration and participants maintenance with web-based interface • Click to connect ease of use • Secured media and signaling encryption • HD quality video and content • Automatic firewall and NAT traversal with ICE, • Share content with other participants* (Only TURN, and STUN support available on Chrome. -
Kandy Link — Webrtc Gateway Using the Web to Extend the Value of Service Provider Networks
Kandy Link — WebRTC Gateway Using the Web to Extend the Value of Service Provider Networks The Impact of WebRTC The adoption of the WebRTC standard by all of the major web browser providers creates an enormous opportunity to change the way people communicate. WebRTC makes it possible to engage in a two-way multi-media conversation (voice, video, screen share) without any extra software or purpose-built devices. Any modern device with a mic and speakers is now a full communications endpoint supporting voice, video and col- laboration tools. The simplicity of WebRTC-based communication access is in sharp contrast to traditional communication tools that require users to load a special app or a plugin for each website or ser- vice they use. This issue has gone from annoying to business impacting as many organizations have heightened security policies that preclude users from adding new applications. And since WebRTC leverages HTML5 and HTTPS, its firewall Multiple Deployment Options Kandy Link is deployed in a subscription based, fully managed friendly, meaning it works almost everywhere. model. Users can choose the operational deployment model Bridging the Digital Divide that best fits their use case: Ribbon’s Kandy Link Gateway offers service providers a simple Public Cloud – as a Service and scalable way to connect different generations of communi- No hardware or capital investment – WebRTC services are cation elements and customer experiences. delivered from the public Kandy cloud with access to Kandy’s entire apps portfolio. • Kandy Link can deliver line or station-side SIP services to existing communication elements (as a service, in network Private Cloud or on-premises) while delivering WebRTC-based services to In network or in a 3rd party data center. -
Audiocodes Webrtc Solution DATA SHEET
AudioCodes WebRTC Solution DATA SHEET Simplified Web Calling for Contact Centers and Service Providers The AudioCodes WebRTC solution is a quick and straightforward way for contact centers and service providers to supply intuitive and high-quality web calling functionality to their service centers. From implementing a simple click-to-call button on a consumer website or mobile app, right up to a fully featured agent client for voice and video calls, the AudioCodes WebRTC solution provides all the building blocks that you need to get web calling services up and running quickly and easily. Simple and scalable Powerful and robust Easy to install A single WebRTC device, either The WebRTC solution is fully integrated The Client SDK facilitates the quick virtual or hardware-based, in AudioCodes Mediant session border integration of the WebRTC solution seamlessly scalable from just a few controllers and inherits the SBCs’ into client websites or Android and sessions to several thousand strong security, high availability and iOS mobile apps transcoding capabilities Signaling AUDIOCODES Browser / Mobile WebRTC SDK WebRTC user Media Internet AudioCodes SBC Integrated WebRTC Gateway Carrier/Contact Center VoIP Network AudioCodes WebRTC Solution DATA SHEET Specifications WebRTC Gateway About AudioCodes AudioCodes Ltd. (NasdaqGS: AUDC) is a Deployment leading vendor of advanced voice networking VMWare KVM AWS Mediant 9000 Mediant 4000 method and media processing solutions for the digital workplace. With a commitment to the human 3,000 voice deeply embedded in its DNA, AudioCodes 5,000 WebRTC sessions 2,700 3,500 (20,000 on 1,000 enables enterprises and service providers (20K on roadmap) roadmap) to build and operate all-IP voice networks for unified communications, contact centers 3,000 1,050 integrated and hosted business services. -
Nanostream Webrtc.Live
nanoStream WebRTC.live Product Overview – Document V 1.4, 2017-01 © 2017 nanocosmos gmbh Setup secure online meetings at a distance or stream live events to thousands of worldwide viewers: nanoStream WebRTC.live makes it very easy to create custom live video communication and interactive live streaming applications with low latency. In combination with the nanoStream H5Live Player, nanoStream WebRTC.live offers a seamless user experience for plugin-free live video broadcast on any device. What is nanoStream WebRTC.live? nanoStream WebRTC.live is a product platform developed by nanocosmos based on WebRTC and other streaming technologies like RTMP and H5Live. It is used in a variety of low-latency interactive live streaming use cases. WebRTC.live video chat combined with RTMP broadcasting allows a unified communications strategy for companies - from 1-on-1 discussions with employees at a distance to sharing live events with 10.000 viewers. In addition to secure peer-to-peer communication, the WebRTC.live encoding platform can be used to broadcast live streams to large audiences. The live video broadcast can be integrated into any live streaming infrastructure with the unique WebRTC-RTMP server bridge, or in an end-to-end development with the nanoStream Cloud. What is WebRTC? WebRTC is a technology standard for audio/video-based real time communication. WebRTC video chats and broadcasts function directly within the web browser, with no additional plugin or app download required. The standard is supported by most browser vendors (Google Chrome, Firefox) and currently available plugin-free on all platforms (Windows, MacOS, Android), except Apple iOS. -
Features Guide [email protected] Table of Contents
Features Guide [email protected] Table of Contents About Us .................................................................................. 3 Make Firefox Yours ............................................................... 4 Privacy and Security ...........................................................10 The Web is the Platform ...................................................11 Developer Tools ..................................................................13 2 About Us About Mozilla Mozilla is a global community with a mission to put the power of the Web in people’s hands. As a nonprofit organization, Mozilla has been a pioneer and advocate for the Web for more than 15 years and is focused on creating open standards that enable innovation and advance the Web as a platform for all. We are committed to delivering choice and control in products that people love and can take across multiple platforms and devices. For more information, visit www.mozilla.org. About Firefox Firefox is the trusted Web browser of choice for half a billion people around the world. At Mozilla, we design Firefox for how you use the Web. We make Firefox completely customizable so you can be in control of creating your best Web experience. Firefox has a streamlined and extremely intuitive design to let you focus on any content, app or website - a perfect balance of simplicity and power. Firefox makes it easy to use the Web the way you want and offers leading privacy and security features to help keep you safe and protect your privacy online. Mozilla continues to move the Web forward by pioneering new open source technologies such as asm.js, Emscripten and WebAPIs. Firefox also has a range of amazing built-in developer tools to provide a friction-free environment for building Web apps and Web content. -
Mivoice Border Gateway Engineering Guidelines
MiVoice Border Gateway Engineering Guidelines January, 2017 Release 9.4 1 NOTICE The information contained in this document is believed to be accurate in all respects but is not warranted by Mitel Networks™ Corporation (MITEL®). The information is subject to change without notice and should not be construed in any way as a commitment by Mitel or any of its affiliates or subsidiaries. Mitel and its affiliates and subsidiaries assume no responsibility for any errors or omissions in this document. Revisions of this document or new editions of it may be issued to incorporate such changes. No part of this document can be reproduced or transmitted in any form or by any means - electronic or mechanical - for any purpose without written permission from Mitel Networks Corporation. Trademarks The trademarks, service marks, logos and graphics (collectively "Trademarks") appearing on Mitel's Internet sites or in its publications are registered and unregistered trademarks of Mitel Networks Corporation (MNC) or its subsidiaries (collectively "Mitel") or others. Use of the Trademarks is prohibited without the express consent from Mitel. Please contact our legal department at [email protected] for additional information. For a list of the worldwide Mitel Networks Corporation registered trademarks, please refer to the website: http://www.mitel.com/trademarks. Product names mentioned in this document may be trademarks of their respective companies and are hereby acknowledged. MBG - Engineering Guidelines Release 9.4 January, 2017 ®,™ Trademark of Mitel Networks -
A Case for SIP in Javascript", IEEE Communications Magazine , Vol
Copyright © IEEE, 2013. This is the author's copy of a paper that appears in IEEE Communications Magazine. Please cite as follows: K.Singh and V.Krishnaswamy, "A case for SIP in JavaScript", IEEE Communications Magazine , Vol. 51, No. 4, April 2013. A Case for SIP in JavaScript Kundan Singh Venkatesh Krishnaswamy IP Communications Department IP Communications Department Avaya Labs Avaya Labs Santa Clara, USA Basking Ridge, USA [email protected] [email protected] Abstract —This paper presents the challenges and compares the applications independent of the specific SIP extensions alternatives to interoperate between the Session Initiation supported in the vendor's gateway. Protocol (SIP)-based systems and the emerging standards for the Web Real-Time Communication (WebRTC). We argue for an We observe that decoupled development across services , end-point and web-focused architecture, and present both sides tools and applications promotes interoperability. Today, the of the SIP in JavaScript approach. Until WebRTC has ubiquitous web developers can build applications independent of a cross-browser availability, we suggest a fall back strategy for web particular service (Internet service provider) or vendor tool developers — detect and use HTML5 if available, otherwise fall (browser, web server). Unfortunately, many existing SIP back to a browser plugin. systems are closely integrated and have overlap among two or more of these categories, e.g., access network provider Keywords- SIP; HTML5; WebRTC; Flash Player; web-based (service) wants to control voice calling (application), or three communication party calling (application) depends on your telephony provider (service). The main motivation of the endpoint approach (also I. INTRODUCTION known as SIP in JavaScript) is to separate the applications In the past, web developers have resorted to browser (various SIP extensions and telephony features) from the plugins such as Flash Player to do audio and video calls in the specific tools (browsers or proxies) or VoIP service providers. -
The Security of Webrtc Ben Feher, Lior Sidi, Asaf Shabtai, Rami Puzis Ben-Gurion University of the Negev {Feherb, Liorsid, Shabtaia, Puzis}@Bgu.Ac.Il
The Security of WebRTC Ben Feher, Lior Sidi, Asaf Shabtai, Rami Puzis Ben-Gurion University of the Negev {feherb, liorsid, shabtaia, puzis}@bgu.ac.il ABSTRACT primarily browser–based (though not all browsers support WebRTC is an API that allows users to share streaming WebRTC) we focus our analysis on WebRTC browser-based information, whether it is text, sound, video or files. It is impelmetations. A quick look at browser usage statistics reveals supported by all major browsers and has a flexible underlying that currently, Chrome and Firefox dominate the browser market infrastructure. In this study we review current WebRTC structure by a significant margin. In addition, these browsers also support and security in the contexts of communication disruption, key features of WebRTC technology which makes them prime modification and eavesdropping. In addition, we examine candidates for our tests. WebRTC security in a few representative scenarios, setting up and Another point for consideration in the security analysis involves simulating real WebRTC environments and attacks. the communication restrictions of the current security measure (e.g., deep packet inspection Firewalls), as well as the availability General Terms of existing tools for analyzing the WebRTC traffic. Security, Network. 2. WHAT IS WebRTC? Keywords In General, a WebRTC communication is composed of two WebRTC, DTLS-SRTP, Signaling, P2P. stages: a signaling and a communication stage. 1. INTRODUCTION 2.1 Signaling Stage Web Real-Time Communication (WebRTC) is an API aimed to The signaling stage is the conversation setup. When two clients support browser-to-browser communication. This solution is wish to communicate, they must arrange a common information designed to unify the fragmented solution cluster that has been channel. -
A 3D Collaborative Editor Using Webgl and Webrtc
A 3D collaborative editor using WebGL and WebRTC Caroline Desprat Jean-Pierre Jessel Hervé Luga [email protected] [email protected] [email protected] VORTEX team, IRIT UMR5505, University of Toulouse, France Online collaborative 3D design Objectives Collaborative design environments share 3D scenes • pluginless visualization of 3D scenes on the web, across the network. 3D scenes or models are very • exclusive use of client resources for likely to be constructed and reviewed by more than communication, one person, particularly in the context of 3D sci- • message broadcasting: asynchronous and entific visualization [1], CAD (Computer Aided De- granular, sign) [2] or BIM (Building Information Modeling) • real-time collaboration across the network with [3] solutions with various architectures [4] [5]. shared access to 3D models. Question What architecture is the best suited in online 3D collaborative design environments for small teams using exclusively browser’s ressources? Our hybrid architecture for 3D web-based collaboration Web-based editor P2P communication Client visualization, interaction and edition. WebRTC and DataChannels The 3D interface integrates a viewport and pro- vides: WebRTC (Web Real-Time Protocol) allows Da- P2P connections be- • the list of contributors, taChannel protocol for tween web brow-sers to exchange any raw • basic transformations tools (CRUD), data in real-time[6]. • and viewport informations. WebRTC uses signaling mechanism (via WebSocket server) to manage peers, the network configuration and the media capabilities. Server: RESTful architecture Auto-management of the P2P full mesh network of REST architecture (client/server/NoSQL DB users (designed for small teams). queries) benefits distributed hypermedia systems. PeerJS library bootstraps WebRTC API for compat- ibility reasons. -
Cisco Remote Expert Mobile 10.6(3) Design Guide
Cisco Remote Expert Mobile 10.6(3) Design Guide First Published: 2015-07-31 Last Updated: 2016-01-13 Cisco Systems, Inc. www.cisco.com 1 Design Guide THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS. THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY. The Cisco implementation of TCP header compression is an adaptation of a program developed by the UniVersity of California, Berkeley (UCB) as Part of UCB’s Public domain version of the UNIX oPerating system. All rights reserVed. CoPyright © 1981, Regents of the UniVersity of California. NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED “AS IS” WITH ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF DEALING, USAGE, OR TRADE PRACTICE. IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. -
Webrtc the PANACEA for INTUITIVE UBIQUITOUS ONLINE COLLABORATION?
IS WebRTC THE PANACEA FOR INTUITIVE UBIQUITOUS ONLINE COLLABORATION? WebRTC (Web Real Time Communications) is a developing open What is WebRTC? standard for enabling web browsers to support multimedia audio, video chat and other data applications. The WebRTC approach to WebRTC is a standard emerging out of the World Wide Web Consortium online real-time communications aims to remove the hassles and (W3C) and the Internet Engineering Task Force (IETF). This standard issues associated with the current practice of installing vendor-specific holds significant promise as a replacement for the current reliance software plug-ins or browser extensions, and promises to vastly on proprietary browser extensions, troublesome plugin downloads improve the user experience. and for potentially eliminating the need for licensed vendor-supplied soft clients. Interoperability complexities and challenges, as well as costs, are a major obstacle for current vendor solutions. Typically, the end-user WebRTC is about putting real-time communications (RTC) capabilities experience suffers, which has led to low adoption and use in stark into a standard browser, with no download, no plugins and no Flash. contradiction to the web browser-based application market. The standard includes echo-cancellation, packet loss concealment, or all functionality expected in a media stack, prevalent in most IP PBX/UC WebRTC shows promise for not only improving the user experience systems or multimedia endpoints. and adoption of real-time communications, but also for unleashing a new wave of video, voice, and data web applications. The WebRTC standard stipulates a framework for browser-to-browser multimedia communications, independent of platform or operating This paper explores some of the technical components for WebRTC system. -
Audio Synthesis and Effects Processing with Web Browser In
Aalto University School of Science Degree Programme in Computer Science and Engineering Jarkko Tuomas Juslin Audio synthesis and effects processing with web browser in open web Master's Thesis Espoo, January 14, 2016 Supervisors: Professor Petri Vuorimaa, Aalto University Advisor: Jari Kleimola D.Sc. (Tech.), Aalto University Aalto University School of Science ABSTRACT OF Degree Programme in Computer Science and Engineering MASTER'S THESIS Author: Jarkko Tuomas Juslin Title: Audio synthesis and effects processing with web browser in open web Date: January 14, 2016 Pages: vii + 47 Major: Department of Media Technology Code: T-110 Supervisors: Professor Petri Vuorimaa Advisor: Jari Kleimola D.Sc. (Tech.) This thesis examines the viability of audio synthesis and effects processing in open web with and within a web browser. Two use cases are presented: local audio synthesis in web browser utilizing PNaCl and Emscripten technologies, and audio synthesis as a service utilizing Web Sockets and WebRTC technologies. Devised test procedures aim to find out the end-to-end audio latency and polyphony performance of the implemented systems. The second use case is also examined from general perspective on distributed audio applications by gained experience while doing this work. After a short introduction the thesis present more thorough background to the subject and describes the related concepts and used technologies. The use cases and test procedures and settings are introduced next along with measured re- sults. The results are analysed and discussed after and finally the conclusions are presented. The lowest latency was achieved with PNaCl measuring 39 milliseconds for real- time audio synthesis in web browser, being on the par with baseline measure- ments with native desktop performance.