Flash or HTML5 that is the question? Piehead has compared the two approaches, and compiled an infographic outlining the pros and cons of each. Overall,  neither Flash nor HTML5 can safely be ignored.

Source: http://www.piehead.com/blog/2012/01/flash-vs-html5

Entering 2012, we are please to announce a new important packaging update of the VXI* VoiceXML browser 6.3 release. First, we will provide more packages for most Linux OS kernels, base on Linux Debian and CentOS/Redhat distributions. Now, you can install VXI* on latest CentOS 6.0, 6.1, 6.2 kernels both 32bit or 64bit. By the way, the previous packages for CentOS 5 have been updated too.

Download all our packages:

• Debian 6.0 Squeeze 32bit | 64bit
• Debian 5.0 Lenny 32bit | 64bit
• CentOs 6.0-6.2 32bit | 64bit
• CentOs 5.4-5.7 32bit | 64bit

More information: http://www.i6net.com/downloads/

The announcement by Adobe last week of the death of Flash on mobile devices will have a significant impact on software development platforms battle for mobiles, but it’s not the end of this war. The way you develop your mobile applications is important but it’s not the only key element to consider for online multimedia services. Our question is now: Is HTML5 able to replace Flash technology for video calling? Obviously, experts working on life video communications over the web, HTML5 is not yet ready but why?

HTML5

There’s two tags to consider for video services <video> and <device> to send and receive a video stream. Unfortunately, these tags are both unable to control in real time a life video/audio streaming. Video calling requires an application to control the bandwidth, decoding/encoding and manage a channels’ real time synchronization.  Currently, most web browsers are not supporting  all <video> formats allowed. By the way, <device> API is not yet implemented, so it’s not possible to use your webcam and microphone with HTML5. This complicated mess in web standards means that you will need to embed separate versions of your video streaming channel to appease the requirements of each browser. This makes impossible to create and deploy any universal video call service thru HTML5 today.

Flash/RTMP

RTMP is available as an open specification to create products and technology. This public protocol is designed for web browsing to be controlled by an cross-platform application: Flash Player or any other application able to manage the public Real Time Messaging Protocol (RTMP). Flash has provided rich camera and microphone universal access for several years now and supports most of multimedia devices. RTMP is omnipresent in the world wide web experience, inside banners, diagrams, infographics, videos, youtube and that’s a great advance because this means almost all web surfers will be able to run your service without any other plugin installation in their computer.

Conclusions:

HTML5 can’t manage a web video call for now. Google, Skype and many others big internet players had tried to define an HTML5 framework to build a web soft phone but they lack of a global solution. Today, Flash players still present in 99% web browsers of the world’s computers while current HTML5 tags are not yet ready for video calling and still not well supported by common web browsers. For mobile devices like iOS or Android, you can build your own native mobile application using RTMP but without Flash Player. So, the winner for web video communications is Flash and its public RTMP protocol for now.

Recommendations:

Our recommendations for video call services is to use the Flash Player for web browsers and create a native application over RTMP for mobiles devices. That’s the best solution in the current situation, and probably for the near future. Otherwise, we will have to reinvent and impose a new standard for multimedia communications, do you like this plan?

“Flash ain’t dead yet” for life video streaming over the Internet and probably for long time.

Sources:

• Youtube: Flash and the HTML5 <video> tag
• BrandFlow – HTML5 Video vs Flash Video
• Wikipedia – Real Time Messaging Protocol
• Adobe – RTMP definition
• Wikipedia – HTML5 

Video Channels:

1. Video h323 Channel
2. Video SIP Channel
3. Video Flash/RTMP Server Channel
4. Video 3G-324m TDM Channel
5. Video Mobile Apps iOS/Android (over RTMP)

1. Video h323 – VoIP

H.323 call signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN. A call model, similar to the ISDN call model, eases the introduction of IP telephony into existing networks of ISDN-based PBX systems, including transitions to IP-based PBXs.

Key Facts:

• Standard based VoIP protocol
• Most hardware conferencing solutions runs h323
• No evolution

Usage:

• To connect some specific video-conferencing or video-phone devices

Trend:

• Today, h323 is an old protocol not easy to configure and to manage. Keep this protocol only for specific devices you have to connect to your IVVR. Try to select SIP, if your device support it for a better service evolution.

2. Video SIP – VoIP

The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams.

Key Facts:

• Large number of VoIP devices and softphones (+)
• Flexible signaling protocol well (+)
• Very few video softphones are available for mobiles devices (-)

Usage:

• To connect video-conferencing or video-phone devices or video-SIP trunks

Trend:

• Any new VoIP device is supporting SIP, it’s the standard protocol for voice and video over IP communications. You cannot avoid SIP to connect your service thru trunks.

3. Video 3G-324m – TDM

The 3G-324M protocol operates over an established circuit switched connection between two communicating peers. 3G-324M is an umbrella specification to enable conversational multimedia communication over Circuit Switched (CS) networks and has been adopted by the 3GPP. 3G-324M is based on the ITU-T H.324 specification for multimedia conferencing over Circuit switched networks.

Key Facts:

• No app. installation required in your mobile phone (+)
• Interoperable thru UMTS network (+)
• Low screen size QCIF and poor video quality limited to 64kbps (-)

Usage:

• To make video calls over the UMTS wireless networks from any 3G videocall compliant mobile phone.

Trend:

• Available in most countries with UMTS networks but wireless carriers haven’t promote enough 3G videocalls and there’s very few services applications available in the market. Anyway, it’s the faster option to make a videocall and make it interoperable for 3G mobiles.

4. Video Flash/RTMP – IP

Real Time Messaging Protocol (RTMP) was initially a proprietary protocol developed by Macromedia for streaming audio, video and data over the Internet, between a Flash player and a server. Macromedia is now owned by Adobe, which has released the specification of the protocol for public use. Flash/RTMP is present in web browsers on 99% of the world’s computers.

Key Facts:

• Flexible bandwidth consumption management for QOS (+)
• Any size QCIF, CIF… HD and video quality available (+)
• Flash dependent, you need a Flash player (-)

Usage:

• To make voice or video calls over any web browser (flash enabled) or any OS.

Trend:

• Any device with a browser will work, this is 99,99% Windows, Mac or Linux computers and most Android Nebooks, Tablets or Smartphones. This solution runs without any addon or executable application.

5. Video iOS /Android – IP (over RTMP)

An iOS or Android mobile application can make videocalls and can be easily customized for many services. This option will allow any iPhone/iPad or Android device to make a call to a video contact center. Your own native mobile application can transcode RTMP format and connect to a remote VXI* server running the Flash/RTMP server channel. No Flash Player is required.

Key Facts:

• You have to install an application in your mobile phone (+)
• Service oriented, each service could use it’s own videophone app. (+)
• No interoperable with other IP telephony services (-)

Usage:

• To make voice or video calls over Internet from iPhone, iPad or Android smartphones.

Trend:

• Mandatory option for any video service for iOS and Android smartphones

Thank you for your continuous support!

(Click to enlarge)

VXI* is the VoiceXML browser for Asterisk and its Xtras* extensions allow to build advanced IVR / IVVR platforms to run standard voice and video telephony applications. This diagram shows each parts of a complete, flexible and evolutive business solution.

Software Components:

• VXI*, VoiceXML browser for Asterisk
• Xtras*, Software extensions for VXI*
• Asterisk, the open source telephony system

Speech engines:

• SIV, Speaker Indentification & Verification (third party)
• TTV, Text-to-Video (internal)
• TTS, Text-to-Speech (third party)
• ASR, Automatic-Speech-Recognition (third party)
• MRCP, Media Resource Control Protocol for ASR (uniMRCP)

More information:

• How to Start
• Documentation
• Hardware servers
• Installation

We present a new update of the VXI* VoiceXML Browser Installation Guide; this new PDF file is now available at our download documents section. You will learn more about how to setup and modify all configuration parameters. This new documentation ref. 07/2011 has been updated for all VXI* 6.0, 6.1, 6.2rc,… releases  and will provide important help to all users.

Download:

• Vxi i6net Installation-Guide-V62.pdf - 723 hits

Some updates:

• System management
• Configuration description (vxml.conf)
• VoiceXML Accounts configuration (accounts)
• Cache management
• Create your first VoiceXML service
• Commercial license activation
• Configuration files (examples)

More information at: Documents

Thank your for your support!

June 20th, 2011, Madrid, Spain. We are pleased to announce the availability of Lumenvox Text-to-Speech (TTS) connector to the VXI* VoiceXML browser for Asterisk platforms. It’s new TTS option but it’s really interesting because Lumenvox is able to provide both ASR and TTS engines for VXI*. In 2008, It was the first Speech Recognition provider for Asterisk and the first one running with the VXI* VoiceXML browser too.

Lumenvox TTS voices are available in: British English, American English, American Spanish with many more coming soon, e.g. French, Australian English, German.

More information about TTS connectors for VXI* here.

Other Posts:

• Lumenvox, Speech Recognition as a Service
• Lumenvox Speech Engine 9.0 for VXI* VoiceXML browser
• VXI* adds MRCP support for ASR…

New VXI* VoiceXML browser  6.1 has been enhanced with new commands line to provide more information to administrators. VXI* data logs are now managed easier from the Asterisk *CLI>; today get online information about all the calls, sessions, accounts, statistics, cache … your IVR/IVVR management becomes simple (for example):

hostname*CLI> vxml show version
Version      : V6.1
Build with   :
CVS Revision : $Revision: 1.469 $
Gcc          : V4.3
Arch         : 32 bits
Target       : i686
Asterisk     : V1.6.2.14
_            : by hudson@debian-50-i386
_            : on i686/Linux 2010-09-06 23:01:26 UTC
Options sum  :
Date         : May 24 2011 10:04:30
hostname*CLI>


Find here the new commands line dumped with Asterisk *CLI>: Help

• vxml cache | Cache actions *New
• vxml debug | Enable VoiceXML application debugging *New
• vxml no debug Disable | VoiceXML application debugging *New
• vxml originate | Originate a call to the VoiceXML application *New
• vxml reload | Reload VoiceXML interpreter configuration
• vxml show accounts | Show the accounts configured
• vxml show account | Show an account configured
• vxml show applications | Show and check the dependencies applications
• vxml show cache | Show files in the cache *New
• vxml show configuration | Show the configuration of VoiceXML interpreter
• vxml show dates | Show dates of VoiceXML interpreter
• vxml show license | Show license of VoiceXML interpreter
• vxml show session | Show session of VoiceXML interpreter
• vxml show sessions | Show sessions of VoiceXML interpreter
• vxml show statistics | Show statitics of VoiceXML interpreter
• vxml show top | Show top of VoiceXML interpreter
• vxml show version | Show version of VoiceXML interpreter

Previous Posts:

• BlogPost: New *CLI> commands for VXI* 4.1
• BlogPost: New VXI* VoiceXML 6.1 browser released!

Build your video IVR menus using SVG format. VXI* VoiceXML Browser 6.0 supports SVG format with the vxml <audio> tag like major web browsers does! With this feature, implementing a nice video menus or adding a wallpaper picture become really easier to design and to update. No pixels resizing, your content will always fit well into your phone video screen.

Scalable Vector Graphics (SVG) is a family of specifications of an XML-based file format for describing two-dimensional vector graphics, both static and dynamic. The SVG specification is an open standard that has been under development by the World Wide Web Consortium (W3C) since 1999. SVG images and their behaviors are defined in XML text files.

SVG Free Editor Tool: InkScape
Open Source SVG editor for many OS (Mac, Linux, Windows)
Website – http://www.inkscape.org

Please contact us for more information

Find here a small example about how to implement a VoiceXML callback feature for your IVR service. VXI* won’t pikup the call of the caller, a ringback tone is playing before handup. Once the call is hangup, the platform callback the caller (caller ID or session.connection.remote.uri). A callback service is a very good idea to avoid call charges for your customers when they are calling you.

account1:

1. Call 0001
2. the IVR pickup the call…
3. and execute your service (charged if you call it directly)

account2:

1. Call 0002
2. the IVR don’t pickup the call… ringbacktone message
3. and call back you to execute your service (free call, with callback)

System Requirements:

• Asterisk 1.4, 1.6, 1.8
• VXI* VoiceXML browser 6.0 or upper
• VoIP or E1/T1 carrier must support ringbacktone/earlymedia

VoiceXML Accounts: vxml.conf

[account1]
name=voicexml_service
number=0001
url=http://localhost/vxml/index.vxml
dialformat=SIP/%s

[account2]
name=notanswer_callback
number=0002
url=http://localhost/vxml/callback.vxml
dialformat=SIP/%s
force=ringing


VoiceXML Code: callback.vxml

Very simple, isn’t it?