Developing WebRTC apps? 
No need to build your own back-end system.

IP Videocom is a scalable communications platform
for real time voice, video
& data WebRTC apps.

Flexible - Scalable - Secure - Low Cost.
Integrates with other types of IP communications.

Launch your WebRTC apps within days, not months.
NO CAPEX.  Runs in the cloud.  Saves time
& money.

Copyright 2012, 2013 IP Videocom
             All Rights Reserved
WebRTC is the New Standard for Real Time Web Communications

WebRTC enables browser-to-browser communications without having first to install plug-ins or client
software.  The WebRTC framework allows developers to add real time functions such as voice calls,
video chats & file sharing directly into web applications.

Industry analysts predict that WebRTC will transform the entire world of IP Communications.  You will no
longer need 3rd-party applications like Skype, Webex, or chat programs.  You can simply embed these
functions directly into your website, turning static web pages into a real time application.

In the near future, WebRTC will become part of every web strategy for enterprises, small business,
social sites, learning institutions, on-line merchants and any customer-facing website in every industry sector.
Development Challenges . . .

You're a Web Developer who's interested in WebRTC.  You've read up on it, attended a seminar, saw some demos & are excited about working with it.  Maybe you want to add a 'Click-to-Call' button to your existing website, or perhaps you want to embed WebRTC functions into a brand new web app that is under construction.  Perhaps you're an entrepreneur who wants to launch a new WebRTC App service, or an enterprise looking to provide sales & support in real time via the browser.

It all looks amazingly simple since it's just JavaScript.  You don't need to write a plug-in since popular browsers already support it.  Then you realize that WebRTC is only a framework and that it doesn't work all by itself.  Even though you may just want to add basic peer-to-peer communications, you will still need to implement some type of signaling protocol in order to establish connections between peers.  You're familiar with a lot of protocol acronyms...SIP, XMPP, Jingle, RTP, SRTP, SDP, STUN, TURN, ICE, etc. but maybe you haven't developed your own code using them.  Then come the questions that you have to find real-world answers to before you can get going:

      •  Which signaling protocol might be best for my particular application?
      •  What about Session Control?  Where should that be implemented?  
      •  How will the promise of peer-to-peer communications work if users are behind a NAT? 
      •  What functions will the back-end server need for my particular WebRTC app?
      •  How will peer-to-peer users know when each is available & ready to connect?  
      •  What about security?  Can we encrypt voice calls & video conferercing sessions using WebRTC?
      •  Will WebRTC apps integrate with our current VoIP-PBX system?  Will it require an upgrade? 
      •  Will the system be able to scale to accommodate hundreds or even thousands of users?
      •  What about compatibility and interoperability between various types of devices?
      •  What about video?  Will I have to use the H.264 codec?  What about transcoding?
      •  Can a TV set-top box become an endpoint?  What about IPTVs?  How about gaming consoles?
      •  What about Federation?  Will my apps be able to communicate with users of other apps? 

Suddenly the project takes on another dimension, and where do you go from here?  There is a solution.


IP Videocom is a scalable platform that not only supports WebRTC, but can be used for practically any type of IP-based voice & video communication from basic Internet telephony to full-featured Unified Communications.  Developed by industry experts with decades of voice, data & video communications experience, the IP Videocom platform can not only take the pain out of your WebRTC development efforts, you can get your WebRTC applications up & running in no time. 

Additionally, the IP Videocom platform provides you with scalable cloud infrastructure so you can start small and grow to support hundreds or even thousands of users on-demand.  You can get started using our APIs to connect to a back-end system that's in the cloud & ready to go.  The platform does all the heavy-lifting including signaling, presence, call routing, peer connections, NAT traversal, etc., and provides everything you need to enable a WebRTC application.  IP Videocom is currently available in its beta release for web developers.  If you would like to learn more about IP Videocom or to get started using the platform to build your WebRTC app, contact us today.  Discuss your requirements with one of our experts and see how the IP Videocom platform can solve your WebRTC development dilemmas.    

The IP Videocom Platform runs in the Crosspeer Cloud.  Crosspeer provides a cloud infrastructure that is OPEN and utilizes KVM, the highest-performing hypervisor available today.  Crosspeer IAAS can support any type of application, but the Crosspeer Cloud architecture was especially designed to support the demanding requirements of today's IP-based communications and to enable applications for the Real Time Web.

With all cloud computing resources independently scalable, true static IP addressing, all storage being persistent & 100% uptime guarantee, Crosspeer Cloud is the perfect choice to deliver a scalable WebRTC solution.  With the IP Videocom platform you can choose to deploy your own 'private' WebRTC solution in the cloud, or utilize the platform as a managed service.  The WebRTC revolution is just getting started and the applications are endless.  We're keeping WebRTC apps open with Crosspeer Cloud.
 
 
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Communicating the Real Time Web