William Guyton Legal Services Alabama I.T. Manager.

Slides:



Advertisements
Similar presentations
What’s New? What’s Different?
Advertisements

SIP, Presence and Instant Messaging
SIP and Instant Messaging. SIP Summit SIP and Instant Messaging What Does Presence Have to Do With SIP? How to Deliver.
Fall IM 2000 Introduction to SIP Jonathan Rosenberg Chief Scientist.
VON Europe /19/00 SIP and the Future of VON Protocols SIP and the Future of VON Protocols: Presence and IM Jonathan Rosenberg.
Fall VoN 2000 SIP for IP Communications Jonathan Rosenberg Chief Scientist.
IETF in the Browser Harald Alvestrand. The Purpose of the IETF The goal of the IETF is to make the Internet work better. The mission of the IETF is to.
Facts about Welcome to this video from Ozeki. In this video I will present what makes Ozeki Phone System XE the Worlds best on-site software PBX for Windows.
XProtect® Web Client 1 Product presentation.
webRTC Andreas Trantidis
UCA Lync Client for Avaya
Analysis of Tools to Support Remote Access to a K12 Classroom John Bowling.
What’s new in this release? September 6, Milestone Systems Confidential Milestone’s September release 2012 XProtect ® Web Client 1 Connect instantly.
Reza hooshangi ( ). short history  One of the last major challenges for the web is to enable human communication via voice and video: Real Time.
10 C H A P T E R © 2001 The McGraw-Hill Companies, Inc. All Rights Reserved1 Streaming Media and Synchronized Multimedia One of the ways the Internet is.
SIP Simplified August 2010 By Dale Anderson. SIP Simplified Session Initiation Protocol Core of SIP specifications is documented in IETF RFC 3261 Many.
Jon R.Doyle VP Business Development CommuniGate Systems.
VCT May 20, 2009 Sapna Blesson Advisor: Dr.Christopher Pollett.
Define objects and their relationships to multimedia Explain the fundamentals of C, C++, Java, JavaScript, JScript, C#, ActiveX and VBScript Discuss security.
UWWD In our quest to eliminate bad websites, we present…. HALLELUJAH!!
VoipNow Presentation Hosted Unified Communications.
Maintain and Modify By: Sahar Aftab (1253 ) and Mehboob Nazim (1085) Central Library.
Media Engineering and Technology 2008 Bachelor Thesis Projects Dr. Fatma Meawad.
1 NETE4631 Communicating with the Cloud and Using Media and Streaming Lecture Notes #14.
Asterisk based web real time communication Advisor : Lian-Jou Tsai Student : Jhe-Yu Wu.
PHP and MySQL Week#1  Course Plan.  Introduction to Dynamic Web Content.  Setting Up Development Server Eng. Mohamed Ahmed Black 1.
Presence Applications in the Real World Patrick Ferriter VP of Product Marketing.
Lightning Talk Fred Rodriguez Aakash Juneja CPSC 473 March 16, 2012.
Berlin, Björn Schwarze.
An Integration Vision IV Ashton. Goals  Helping as many people as we can with limited resources  Connecting People that Need Resources to those that.
Lightning Talk Fred Rodriguez Nguyen Do CPSC 473 May 6, 2012.
WebRTC Multimedia in www Ján Murányi, Ivan Kotuliak.
VoIP Voice over Internet Protocol H.323 SIP RTP SDP IAX SRTP Skype And a lot more…
TSMN 6350 IP TELEPHONY Class Project Mentor: Aishwarya Srinivasan – Team: Monisha Yerramalla –
CIS 1310 – HTML & CSS 1 Introduction to the Internet.
PackLet A web-based text messaging application using AX.25 packet radio technology.
Asterisk based real-time social chat Advisor : Lian-Jou Tsai Student : Jhe-Yu Wu.
Asterisk based web real time communication Advisor : Lian-Jou Tsai Student : Jhe-Yu Wu.
Computer and Information Science Ch1.3 Computer Networking Ch1.3 Computer Networking Chapter 1.
NETWORK HARDWARE AND SOFTWARE MR ROSS UNIT 3 IT APPLICATIONS.
The Internet CSC September 30, History of the Internet Developed for secure military communications Evolved from Advanced Research Projects.
INTERNET. Objectives Explain the origin of the Internet and describe how the Internet works. Explain the difference between the World Wide Web and the.
Jan 24, 06William Rich, Pingtel Corp. IT Expo. Pingtel Corp., William Rich, IT Expo, Jan 24, 06 VoIP is Here… Source: IDC IP PBX vs. TDM PBX.
Web-based Enterprise Telephony Application Development Johnny Wong Principal Member of Technical Staff Oracle Corporation.
Lonce Wyse Arts and Creativity Lab National University of Singapore Net-Music 2013: The Internet as Creative Resource in Music.
Bridging Two Worlds Parting Is Such Sweet Sorrow: Adding IP Telephony to Existing "Big Iron" Mike Robinson CTO
Submitted By: Aayush Beri Di Wen.  Library : Java Media Framework (JMF)  Protocol and System Design  Code Snippet  Simulation.
Why need IP telephony? Economic (uses internet, IP routers...) Basic packages run around 10-15$ and don’t include national long distance calling Traditional.
1 WebRTC in the Call Center and Number Replacement © 2015 Ingate Systems AB Prepared for:Ingate SIP Trunking, UC and WebRTC Seminars WebRTC in the.
Contents : What is Silverlight? Silverlight Overview Silverlight Toolkit Overview Timeline & Packaging Silverlight V1.0 & V1.1 Properties of V1.0 Properties.
Video Chat – getting ready Need a broadband connection. Need speakers & web cam or headset and webcam. Download software or browser plug-in. Sign up for.
Avaya Communicator for Web Demo Installation
WebRTC Don McGregor Research Associate MOVES Institute
Copyright © 2002 Pearson Education, Inc. Slide 3-1 Internet II A consortium of more than 180 universities, government agencies, and private businesses.
0 What Does SIP Bring to Your Customer Experience ? Extend VoIP and IP Contact Center values through support of SIP o Media and location independent support.
Cisco Confidential 1 C © 2013 Cisco and/or its affiliates. All rights reserved. Revision: Draft 3 September 2013.
Esna Cloudlink 5.0 for Cisco Integrate Cisco Collaboration with Business Applications.
Keith Telle Lead Software Engineer Bit Wizards Behind the Magic: SignalR Demystified.
How to Use Safe Money in Kaspersky? Help Desk Number.
ESNA CLOUDLINK 4.0 Integration
Voice over internet protocol
WebRTC enabled multimedia conferencing and collaboration solution
Browsers and Web Platforms
ESNA CLOUDLINK 4.0 Integration
5 things you didn’t know you can BUILD with Microsoft Edge
WebRTC for Bria Khris Kendrick
Apache Cordova What is it ? Platforms Development Architecture Plugins
Unified Real-Time Communications with Pàdé
WebRTC From Zero to Hero The Rolling Scopes, Gabriel Mičko.
Presentation transcript:

William Guyton Legal Services Alabama I.T. Manager

How do we deliver on IV’s vision? A vision of an integrated delivery model. A vision that integrates business logic and process With communications technology webRTC + asterisk +legal server

So what is webRTC? WebRTC is a free, open project that enables web browsers with Real-Time Communications (RTC) capabilities via simple Javascript APIs.

Enable realtime Communication in the browser (no plugin) Run rich realtime media without extra software (no plugin) Run on existing supported browsers (Chrome and FireFox)

Is now adopted by the internet task force IETF and the W3C consortium A HTML5 standard Supporting different types of media such as jingle/SIP/XMPP Audio, video, telepresence, chat, etc…

Instantly VoIP enabling every browser in the world No software or plug in to install webRTC rocks!

Interoperability with existing VoIP technologies using RTP (SIP) Open Standards: anyone can play, WC3, True unified communications Voice/Video/Screen Sharing/IM

OS agnostic, Linux, Mac, Windows. Network friendly via port 80 and 443 allows for Integrated browser/Internet functions So webRTC is the “client” side of the integration….

Server side….Asterisk Asterisk is a Communications Engine Free, open source, large active community Suport for many protocols and projects…. VoIP, PSTN, fax, IM, call center via queues, many more…..

Asterisk version 11 5 years of support a LTS Release LTS = Stability NOT Features New in Asterisk 11

Chan_motif – combines chan_google and chan_jingle into a single driver Refactored XMPP engine res_xmpp More Stable / less difficult to keep up with google

WebSockets support for SIP WebRTC support, adds real-time communications to web browsers Intergrated into onboard HTTP server in *

Client side via webRTC and a browser Server side via asterisk (communications engine) CMS side via Legal Server So the case management system is integrated with the communication engine and thus aware of communications related events.

So in IV’s call center example: Online intake is integrated with CMS CMS is integrated with call center queues Based on client ID, given after online intake is done, client is re-queued back in line.

So in IV’s call volume example: A business process is triggered base on a number of callers on hold in the hot line queue. An /IM/SMS is generated and send to current volunteers with a imbedded hyperlink that once clicked logs them into the queue to take a call.

The browser is the VoIP phone (SIP) The browser is the IM to program staff and/or client (XMPP) The browser is the screen sharing platform to help a client fill out a form or intake We are currently doing integration work in Alabama (LSA) and Montana (MTLSA).

Thank you Effective and efficient

William Guyton Legal Services Alabama I.T. Manager (334)