WebRTC and Telecommunications
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WebRTC is a Google created set of standards for enabling realtime communications in a browser. More and more companies are looking towards WebRTC enabled applications enabling them to have voice, video and messaging just a login away, no matter the browser, no matter the operating system. This presentation is by AVOXI with guest Ben Klang from Mojo Lingo, a premier telephony services firm in Atlanta, GA.
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Html5 RTC - 1
This document provides an overview of HTML5 Real Time Communication (RTC) and related APIs. It begins with an introduction to HTML5 RTC and its support for real-time applications like voice/video calling and file sharing directly in the browser. It then covers the HTML5 overview, benefits of RTC, relevant JavaScript APIs including RTCPeerConnection, RTCDataChannel and MediaStream/getUserMedia, and outlines a planned workshop to build an RTC application.
Webinar WebRTC HTML5 (english)
WebRTC will enable real-time communications like voice and video directly in web browsers without plugins. The presenters will discuss their vision for this technology and how to implement it for corporations and telecom networks. They will cover introductions to HTML5, WebRTC, and network architectures; technical challenges around codecs, encryption, and NAT traversal; application cases for telecoms, companies, social media, and manufacturers; and demos of WebRTC applications and identity management. The presentation aims to show how voice traffic will migrate to the web, with browsers as new endpoints and websites as potential call centers, changing how telephone numbers and communications are managed.
Видео+Конференция 2015: Секреты WebRTC: как вендоры извлекают пользу из проры...
WebRTC offers real-time communication capabilities directly within web browsers through JavaScript APIs. It is both an open-source software stack and a browser standard specification being developed by the IETF and W3C. The document discusses how different companies can implement WebRTC in various ways such as federating networks, operating standalone services, aggregating users, or embedding widgets. Overall, WebRTC is a technology that must be built upon rather than a complete solution itself.
Видео+Конференция 2015: Секреты WebRTC: как вендоры извлекают пользу из проры...
WebRTC offers real-time communication capabilities directly within web browsers through JavaScript APIs. It is both an open-source software stack and a browser standard specification being developed by the IETF and W3C. The document discusses how different companies can implement WebRTC in various ways such as federating networks, operating standalone services, aggregating users, or embedding widgets. Overall, WebRTC is a technology that must be built upon rather than a complete solution on its own.
WebRTC Workshop 2013 given at the IMS World Forum
The document provides an agenda for a WebRTC workshop covering the following key points:
- The workshop will provide a deep technical and business overview of WebRTC through presentations and demonstrations.
- Attendees will learn about the current status of WebRTC standardization and implementations, and what capabilities may emerge over the next 1-2 years.
- The workshop includes sessions on technology details like APIs, media protocols, and interoperability, as well as implications for service providers, enterprises, and use cases.
- An afternoon demo session will provide hands-on experiences of WebRTC applications from various companies and allow networking among attendees.
WebRTC standards update (13 Nov 2013)
Each WebRTC deployment is implementing its own proprietary signaling mechanism. WebRTC signaling and media is incompatible with existing VoIP deployments, requiring gateways to bridge the two. The WebRTC API specification is still evolving and may take different forms across implementations. Standards bodies are working to define signaling interoperability and bridge WebRTC with existing networks like SIP, IMS and videoconferencing systems.
DevCon5 (July 2014) - Intro to WebRTC
WebRTC provides a standardized profile for real-time communication that enables interoperability between browsers without plugins. It defines client-side APIs for audio and video calling as well as other real-time communication capabilities. The WebRTC architecture includes the API, codecs, transport mechanisms like STUN and TURN, and network I/O that allow real-time apps to run directly in browsers. Signaling is required to establish connections between users, and the standardization of WebRTC aims to improve interoperability compared to proprietary solutions. However, interoperability is not always in the best interests of businesses. Ultimately, the API is more important than the underlying protocols it uses.
Status of WebRTC across Asia by Alan Quayle +++
Status of WebRTC across Asia by Alan Quayle, and a group of leading experts contributing to the reality, not the hype, of WebRTC.
It’s 2020, WebRTC (Web Real Time Communications) became known in 2011 when Google open sourced intellectual property it had bought in previous years. Gossip about those acquisitions began in 2009. The IETF (Internet Engineering Task Force) was already laying the groundwork with Opus (voice codec) officially in 2010, and back in 2009 the discussion process started that became WebRTC. It’s been roughly one decade. Did WebRTC change everything? Is WebRTC everywhere?
WebRTC myths and misconceptions. Understanding the two components of WebRTC, the open source project, and the standards track.
Reviewing the achievements of WebRTC across Asia.
Understanding why ‘WebRTC’ companies such as Vidyo and Tokbox did not achieve big exits.
What is the current status of WebRTC, where are the standards, where is the innovation edge?
What is happening across Asia on WebRTC? Understanding the difference service providers adoption of WebRTC. Across telcos, CPaaS, UCaaS. CCaaS, in-app communication platforms, and enterprises.
Case studies on WebRTC implementation across Asia.
Recommendations for WebRTC in Asia.
Lisboa WebRTC - May 21, 2015 - Intro to WebRTC
WebRTC allows for real-time communication directly in the browser through APIs for video and audio without plugins. It is specified by the W3C and brings capabilities like peer-to-peer connections to the open web platform. While it enables new possibilities, implementing WebRTC still involves challenges related to communications engineering like handling full meshes, bridges, and signaling between different browser implementations. Resources are available to help developers get started with WebRTC and abstract some of its complexity.
ISM APAC TechTalk - WebRTC - Dimuthu Nilanka
WebRTC is an open-source project from Google that allows real-time communication like voice calls, video chat, and file sharing directly in the browser without plugins. It uses APIs to handle media capture, streaming and peer-to-peer data transmission between browsers. Supported browsers include Chrome, Firefox, Opera, and Android versions of Chrome and Firefox.
Is Webrtc Transforming Collaboration
WebRTC is a newly proposed open standard, collection of open source libraries and accompanying network services. How is this technology being leveraged to enable rich collaboration? Originally presented at Interop May 4 2016.
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Видео+Конференция 2015: Секреты WebRTC: как вендоры извлекают пользу из проры...
Видео+Конференция 2015: Секреты WebRTC: как вендоры извлекают пользу из проры...
Видео+Конференция 2015: Секреты WebRTC: как вендоры извлекают пользу из проры...
Видео+Конференция 2015: Секреты WebRTC: как вендоры извлекают пользу из проры...
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WebRTC and Telecommunications
- 1. WebRTC and Telecommunications Will this be your phone in the near future? Steve Youngblood, AVOXI (www.avoxi.com) Ben Klang, Mojo Lingo (www.mojolingo.com)
- 2. What is WebRTC? “WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs.” http://www.webrtc.org/
- 3. What is WebRTC? ● Released by Google in 2011 as an open source project for browser based real-time communications ● Has been followed by IETF efforts to standardize relevant protocols and W3C efforts to standardize browser APIs ● SIP over Websockets is often used due to the applicability of SIP in most communications scenarios
- 4. Primary WebRTC Components ● getUserMedia, which allows a web browser to access the camera and microphone and to capture media ● RTCPeerConnection, which sets up audio/video calls ● RTCDataChannel, which allow browsers to share data via peer-to-peer
- 5. Why WebRTC? ● Enables audio, video and messaging between users with no plugins ● Enables audio and messaging between browsers and other commercial communications environments ● Utilizes greater security found in most of today’s communications systems ● The most simple link to VOIP/SIP ● It’s here today
- 6. Who’s Doing It? ● Twilio’s WebRTC APIs ● Amazon with it’s Mayday video service ● American Express’ Live Video Chat ● Google Hangouts ● Snapchat for voice ● JITSI’s open source web collaboration application
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