Google and Avaya to bring Chromebooks and WebRTC to call centers

Google and Avaya are chasing companies seeking to install or upgrade call center systems, promising them easier and more affordable deployments via Chromebooks and a WebRTC interface to the Avaya customer support software.

The integrated systems would let customer-support reps access the Avaya call center software—hosted in the cloud or in a local network—via the Chrome browser. Tapping the Chrome browser’s native support of WebRTC, users will be able to communicate with audio and video without having to install additional software.

“As we move to omni-channel support, incorporation of video is essential for contact centers,” said Joe Manuele, a vice president at Avaya.

WebRTC technology, heavily supported and promoted by Google, allows audio and video conferencing applications to run on browsers via Javascript APIs without needing special plugins or add-ons.

For example, it will be possible for contact center staffers to launch one-to-many video conference sessions, and to route them to another agent while in progress without interrupting the broadcast, according to Manuele.

Avaya hopes the offering will attract new clients and prompt customers of its call center software to upgrade their systems, especially those looking to move away from traditional Windows desktop PCs loaded with local software or to replace thin, virtualized clients, and thus simplify their infrastructure.

Meanwhile, Google expects the partnership to help spur demand for Chromebook devices. Although Google doesn’t make Chromebooks, the company generates revenue licensing and IT administration software for the devices.

The Avaya-Google bundle includes the Avaya Agent for Chrome software, and the Google Chrome management console. It can be ordered now and will ship in the coming weeks. It’s expected to start at $30 per concurrent user, per month, for a three-year subscription, or $900 for a perpetual license.

Chromebook devices are sold separately. The partnership currently doesn’t include a specific deal with any hardware vendor, but any Google-authorized Chrome device—Chromebook laptops and Chromebox desktops alike—will work with the call center software. Hardware makers that market Chrome devices include HP, Lenovo, Asus, Acer and Toshiba. “There’s a great set of Chrome devices out there for the enterprise,” said Rajen Sheth, Director of Product Management, Chrome for Business and Education at Google.

One customer with concrete plans to adopt the Avaya Agent for Chrome is MeadWestvaco, a logistics company with customers in 100 countries that, according to Avaya, wants to simplify and modernize its contact center systems.

Introduction to WebRTC architecture

What is ICE?

Interactive Connectivity Establishment (ICE) is a framework to allow your web browser to connect with peers. There are many reasons why a straight up connection from Peer A to Peer B simply won’t work. It needs to bypass firewalls that would prevent opening connections, give you a unique address if like most situations your device doesn’t have a public IP address, and relay data through a server if your router doesn’t allow you to directly connect with peers. ICE uses some of the following techniques described below to achieve this:

What is STUN?

Session Traversal Utilities for NAT (STUN) (acronym within an acronym) is a protocol to discover your public address and determine any restrictions in your router that would prevent a direct connection with a peer.

The client will send a request to a STUN server on the internet who will reply with the client’s public address and whether or not the client is accessible behind the router’s NAT.

What is NAT?

Network Address Translation (NAT) is used to give your device a public IP address. A router will have a public IP address and every device connected to the router will have a private IP address. Requests will be translated from the device’s private IP to the router’s public IP with a unique port. That way you don’t need a unique public IP for each device but can still be discovered on the internet.

Some routers will have restrictions on who can connect to devices on the network. This can mean that even though we have the public IP address found by the STUN server, not anyone can create a connection. In this situation we need to turn to TURN.

What is TURN?

Some routers using NAT employ a restriction called ‘Symmetric NAT’. This means the router will only accept connections from peers you’ve previously connected to.

Traversal Using Relays around NAT (TURN) is meant to bypass the Symmetric NAT restriction by opening a connection with a TURN server and relaying all information through that server. You would create a connection with a TURN server and tell all peers to send packets to the server which will then be forwarded to you. This obviously comes with some overhead so is only used if there are no other alternatives.

What is SDP?

Session Description Protocol (SDP) is a standard for describing the multimedia content of the connection such as resolution, formats, codecs, encryption, etc so that both peers can understand each other once the data is transferring. This is not the media itself but more the metadata.

What is an Offer/Answer and Signal Channel?

Unfortunately WebRTC can’t create connections without some sort of server in the middle. We call this the Signal Channel. It’s any sort of channel of communication to exchange information before setting up a connection, whether by email, post card or a carrier pigeon... it’s up to you.

The information we need to exchange is the Offer and Answer which just contains the SDP mentioned above.

Peer A who will be the initiator of the connection, will create an Offer. They will then send this offer to Peer B using the chosen signal channel. Peer B will receive the Offer from the signal channel and create an Answer. They will then send this back to Peer A along the signal channel.

What is an ICE candidate?

As well as exchanging information about the media (discussed above in Offer/Answer and SDP), peers must exchange information about the network connection. This is known as an ICE candidate and details the available methods the peer is able to communicate (directly or through a TURN server).

The entire exchange in a complicated diagram

With Webrtc, the Skype's probably won't the utmost

Analysts expect WebRTC will be a market worth $4.7 billion by 2018

WebRTC, a free browser-based technology, looks set to change the way we communicate and collaborate, up-ending telecoms firms, online chat services like Skype and WhatsApp and remote conferencing on WebEx.

Web Real-Time Communication is a proposed Internet standard that would make audio and video as seamless as browsing text and images is now. Installed as part of the browser, video chatting is just a click away – with no need to download an app or register for a service.

WebRTC allows anyone to embed real-time voice, data and video communications into browsers, programs - more or less anything with a chip inside.

 Already, you can use a WebRTC-compatible browser like Mozilla's Firefox to start a video call just by sending someone a link.

Further ahead, WebRTC could add video and audio into all kinds of products and services, from GoPro cameras and educational software to ATMs and augmented reality glasses. Imagine, for example, wanting to buy flowers online and being able, at a click, to have the florist display arrangements to you live via a video link.

WebRTC will be a market worth $4.7 billion by 2018, predicts Smiths Point Analytics, a consultancy. Dean Bubley, a UK-based consultant, reckons over 2 billion people will be using WebRTC by 2019, some 60 percent of the likely Internet population.

How to start chatting with webRTC, the no-hassle, in-browser voice and video tech

There's a relatively new technology built in to most browsers that could revolutionize the way you talk with your friends and family. Called webRTC, the HTML 5-based tech could one day replace the need for third-party plugins from services like Google Hangouts or Skype, offering voice and video chat capabilities natively in your browser.

Even better, most implementations of the technology don't require an account of any kind. Chats take place on a web page that you set up on a site that supports webRTC. To get chatting all you have to do is share a link to the web page and you'll be up and running in no time. Talk about hassle free!

If you'd like to give webRTC a try, here's how to get started.

Your browser

Many current browsers for the PC support webRTC, including Chrome, Firefox, and Opera. Apple's Safari doesn't and neither does Microsoft's Internet Explorer; however, Microsoft-owned Skype plans on supporting webRTC in the future with its newly announced Skype for Web project.

If you want to go mobile Chrome, Firefox, and Opera support webRTC on Android. On iOS you can try Bowser.

Web Real-Time Communications

Web Real-Time Communications

WebRTC, otherwise known as Web Real-Time Communications is an open-source project – promoted by Google, Mozilla and others – enables plugin-free Real Time Communications via Javascript API’s. It facilitates browser to browser applications for voice calling, video chat and file sharing. The supported codec for WebRTC is currently VP8. WebRTC uses a server called Web Conferencing Server that in conjunction with a STUN Server  it is required to provide the initial page and synchronise the connections between two WebRTC endpoints.

Why is WebRTC important?

The WebRTC project is incredibly important as it marks the first time that a powerful real-time communications (RTC) standard has been open sourced for public consumption. It opens the door for a new wave of RTC web applications that will change the way we communicate today.

Significantly better video quality	WebRTC video quality is noticeably better than Flash.
Up to 6x faster connection times	Using JavaScript WebSockets, also an HTML5 standard, improves session connection times and accelerates delivery of other Gothuo events.
Reduced audio/video latency	WebRTC offers significant improvements in latency through WebRTC, enabling more natural and effortless conversations.
Freedom from Flash	With WebRTC and JavaScript WebSockets, you no longer need to rely on Flash for browser-based RTC.
Native HTML5 elements	Customize the look and feel and work with video like you would any other element on a web page with the new video tag in HTML5

Here's the video from the WebRTC session at Google I/O 2013, which gives a good look at the current state of the WebRTC universe.

WebRTC  Will Explore Changes in Enterprise Communications

• Enterprise UC
• Contact Centers/Customer Service
• Adding Video to Web Apps
• Vertical Applications
• Healthcare
• Financial Services
• Retail