In the Dialogic HMP Program Part – I, I posted about installing and configuring HMP Software and testing it using freely available SoftPhone Software Kapanga. Once one completes that, one eliminates any doubt on the proper installation and configuration of the HMP software and safely start debugging his self written code.

Another important information is that, the code which is written for DNI boards using SS7 protocol ( for example SPCI4 and DNI240 board) using Global Call Protocol, would work as it is, except few steps like changing of the Device Name while opening it with gc_OpenEx() function. After changing these, the source code should work fine for VOIP calls!

Here are the few steps to code HMP :-

1. Start GC protocol using gc_Start() function.
2. Search for available VOIP Devices as well as Voice Resources. I normally enter them manually using a configuration files ( a TEXT file) as it is easier and help in various ways in future!
3. Open voice device using dx_open() function and store the voice handle.
4. Open VOIP Device using gc_OpenEx() function.
5. Add CODEC using gc_SetUserInfo() function;
6. Connect voice device and VoIP device using dx_Listen and gc_Listen() function.
7. Repeat #3 through #5 for all available devices;
8. Finally perform the final IVR call flow as well as other functionality.

The best way to find out working sample source code is to check basic gc_basic_call_model.c file which is found in Dialogic HMP directory demos\gc_basic_call_model. The only problem with this code is, it does not play any WAV/VOX file, only receives call, make calls, disconnects call etc. But this sample code is very useful to learn IVR programming using any type of Dialogic CTI boards.

I will update this post with working sample source code soon!

Recently, I have started getting mails about asking open source IVR as well as its availability for free, download-able from internet! Most of the people who have asked me about “‘Open Source IVR” have heard about Asterisk. Before going into the details about Asterisk, being open source IVR, I would like to clarify few things about IVR ( Interactive Voice Response)  System.

IVR system consists of  four components, one can say. They are :-

1. CTI ( Computer Telephony) board
2. Call Flow development as per requirement
3. Voice prompts, TTS ( Text To Speech) Engine  and ASR (Automatic Speech Recognition) Engine
4. Host PC or Server for #1 through #3 above.

1. CTI ( Computer Telephony) board

CTI board is a hardware and I m quite sure, nobody will make it ‘open source’ or free for anyone! The other alternative is SIP ( Session Initiated Protocol) on VoIP ( Voice Over Internet Protocol). But for connecting to PSTN ( Public Switched Network), TDM based CTI Board is a must and there is no alternative, as of now!

All CTI boards come with their own set of API ( Application Programming Interfaces) to control telecom protocols and call handling. Dialogic, the biggest CTI company provides many APIs as well as sample codes in three major operating systems, Windows, Linux and MAC OS. Likewise, other CTI vendors too provide their proprietary C++ API for at least two OS.

2. Call Flow development as per requirement

There are many online as well as offline call flow development tools. Some Call Flow development tools are completely pictorial visual tool where one can drag and drop various call flow like components to make a call flow. Some others use simple ‘English’ like/natural language like scripts to define call flow. Incidentally, all these tools are free to use for developing the call flow, but to execute these call flow, one has to pay per port licenses.

VXML is standard call flow development language adopted by W3 which uses HTML like simple syntax. Presently this requires a third party interpreter which again requires per port licenses.

3. Voice prompts, TTS ( Text To Speech) Engine and ASR (Automatic Speech Recognition) Engine

Voice prompts are normally recorded in a studio by professional artists. One can also record voice prompts using computer/laptop and MIC for development purpose. There are many open source, freely available TTS. Even MS SAPI is freely available. eSpeak, a popular TTS which has support for many languages from all around the world, is an open source TTS which works well both in Windows and Linux.
Likewise there are many open source ASR that works well after refinements.

Now, major challenge is to integrate all these components and build a reliable IVR system. Now a days, IVR is no longer simple call flow of yes and no by the caller, but complex system with data retrieving from multiple sources. It performs complex task as well compute input data by caller, make a decision and respond to caller with appropriate answer.

Most of the time, making an informed choice of IVR which is most suitable for a particular IVR application is quite difficult.

Now finally, Asterisk is a truly open source telephony system with IVR facility. It is Linux based and supports many CTI hardware and many developers contributing to it. It is a completely DIY ( Do It Yourself) IVR platform. It may not be easy to develop an IVR system quickly for a beginner using Asterisk from a scratch. At the end, it may be as difficult as building IVR using native API available with CTI board itself.

Many high density and widely deployed CTI hardware are not supported by Asterisk. This is one disadvantage which Asterisk will take some time to overcome. And at the end, while some one tries to customise Asterisk, he might end up with his own proprietary piece of codes, specific to some CTI hardware only!

Advantage of using API of CTI board is the freedom of choice of operating system as well as programming language. Programming languages like C++ offer power of ‘real programming’ language to complete complex tasks as well as integrate with any third party software.

So, one may not have much choice as Open Source IVR!

Apex Voice Communication has announced on the sideline of Mobile World Congress, support for Diameter protocol, successor to Radius protocol today in Barcelona, Spain. Supported in the OmniVox3D SIP Application Server and accessed through its OmniView(R) Service Creation Environment, the new Diameter Command Icon enables online charging, including Authentication, Authorization and Accounting (AAA) functionalities.

OmniVox3D now supports Diameter Protocol

Omnivox3D is flagship product from Apex Voice Communications which is Service Delivery Platforms (SDPs) and Application Servers for voice and video enhanced services.

“With support for Diameter in OmniVox3D, network operators around the world can now easily implement all types of online charging,” said Elhum Vahdat, executive vice president of APEX Voice Communications. “Additionally, the new Diameter Command has extensive support for vendor-specific attributes, which we know differs among the various back-end charging and billing systems.”, says the press release.

Here is the press release from Apex Voice Communication :-

APEX VOICE COMMUNICATIONS ANNOUNCES SUPPORT FOR THE DIAMETER PROTOCOL

– AAA Now a Standard Component of OmniVox3D(R) SIP Application Server –

MOBILE WORLD CONGRESS 2010, BARCELONA, SPAIN, February 15, 2010 –
APEX Voice Communications (www.apexvoice.com), a global provider of Service Delivery Platforms (SDPs) and Application Servers for voice and video enhanced services, today announced support for the Diameter Protocol. Supported in the OmniVox3D SIP Application Server and accessed through its OmniView(R) Service Creation Environment, the new Diameter Command Icon enables online charging, including Authentication, Authorization and Accounting (AAA) functionalities. Diameter is the successor to the RADIUS protocol for use in SIP/IMS networks.

The new Diameter Command in OmniVox3D enables online charging using the Diameter Credit Control Application. This Command can be embedded into call dialogs using OmniView, where application developers can create services that can be charged directly to subscribers’ accounts. The incremental nature of the Diameter online charging application permits subscribers’ accounts to be charged for several services simultaneously, enabling enhanced services (such as video/multimedia content, ringtones, etc.) to be charged by an application server while a switch or CSCF/proxy bills the account for the entire call.

“With support for Diameter in OmniVox3D, network operators around the world can now easily implement all types of online charging,” said Elhum Vahdat, executive vice president of APEX Voice Communications. “Additionally, the new Diameter Command has extensive support for vendor-specific attributes, which we know differs among the various back-end charging and billing systems.”

Diameter is the protocol chosen by 3GPP for use in SIP/IMS networks, and has become the protocol of choice for interfacing 4th-generation carrier network equipment to convergent billing systems. Many new subscriber charging and billing systems for network equipment (e.g. from Ericsson, Huawei) use the Diameter Credit Control Application (Ro) as described in RFC 4006 and in 3GPP TS 32.299. Offering better overall transport, proxying, session control and security for Authentication, Authorization and Accounting (AAA) functionalities,
the Diameter Protocol is an important integration point for carrier tracking of their customers’ usage and as the basis for monthly billing.

The OmniVox3D SIP Application Server is the core component of the APEX Service Delivery Platform (APEX SDP[tm]), which is a mature, field-proven and future-proof platform that delivers immediate returns on the new generation of voice, video and data enhanced services. The APEX SDPs inherent design ensures the scalability, reliability and high-availability required by mobile and fixed-line Network Operators and VASPs, while its award-winning OmniVox3D and OmniView offer the flexibility for rapid deployment of service bundles across multiple networks – 3G, SIP/IMS, TDM/SS7, AIN/IN and converged. The APEX SDP and OmniVox3D together can process an almost unlimited number of applications, including converged Web 2.0 and Telecom services (mashups), Network IVR, Video IVR, Messaging, Real-Time Billing (Prepaid), Conference Calling and Automated Collect Calling.

About APEX Voice Communications
(www.apexvoice.com, sales@apexvoice.com, +1 818.379.8400)
APEX is a global provider of Service Delivery Platforms and SIP Application Servers to Network Operators, Value Added Service Providers (VASPs) and Value Added Resellers (VARs) for voice and video enhanced services. With over 15,000 installations across 95 countries since 1989, APEX continues to be a leader in delivering the latest solutions for mission-critical and revenue-generating services.

For more information contact:

Elhum Vahdat
APEX Voice Communications
+1 818.379.8400 x104
evahdat@apexvoice.com

——–
OmniVox3D is a registered trademark and APEX SDP is a trademark of APEX Voice Communications, Inc. Other trademarks are the property of their respective companies.

I had posted a report few weeks back about IVR market growing to $2.7 billion by 2011 provided by DMG Consulting LLC, and today I had come across another press release while Googling, which says IVR market will grow to $514million from present estimate of $431million this year, as per T3i Group LLC, which “provides market research, data, analysis, and consulting and advisory services to the telecommunications industry.”

Few Key findings of the report says :
1. 95% of IVR shipped would be supporting VXML from recent less than 75%.
2. Top three IVR systems are
a. Inbound Call handling for contact centers
b. Inbound IVR self service transactions
c. Outbound IVR for alerts like appointment confirmations, payment collection remainder.
3. Inbound IVR will see decrease by 10% by 2013.
4. Outbound IVR will grow by 12% yearly.
5. Analog CTI boards are decreasing and DTMF based menu in IVR application are also on the decline. Speech enabled menu is growing and will be have 2:1 advantage over DTMF.

Complete Press release, Please visit : http://telecomweb.com/aboutus/pressreleases/262963.html

About T3i Group LLC

T3i Group LLC provides market research, data, analysis, and consulting and advisory services to the telecommunications industry. It has clients in 46 countries and conducts its business through four operating units: InfoTrack publishes reports that analyze shipment, revenue, market share and other pertinent data of importance to telecommunications equipment manufacturers; TelecomTactics maintains a database of the features and functionality of major telephony systems; Tarifica maintains a database of pricing-related information covering 400 telecommunications carriers operating in 130 countries; and TelecomWeb (www.t3igroup.com) serves as T3i Group’s primary delivery mechanism for distributing its reports, analyses and data to subscribers.

T3i Group LLC is headquartered in Parsippany, N.J., with additional offices in New York City; London, England; and Cherry Hill, N.J.

What is Asterisk ?

Asterisk is the world’s leading open source telephony engine and tool kit. Offering flexibility unheard of in the world of proprietary communications, Asterisk empowers developers and integrators to create advanced communication solutions…for free.
Asterisk® is released as open source under the GNU General Public License (GPL), and it is available for download free of charge. Asterisk® is the most popular open source software available, with the Asterisk Community being the top influencer in VoIP. Source : www.asterisk.org

In other words, you can create IVR application using Asterisk as a platform! Do they support Dialogic cards ? They support some Dialogic card but they are mainly supporter of Digium cards.

So, basically Asterisk is a linux based platform to develop CTI Applications right from low end Analog Phones to high end VoIP. They support host of protocols like their paid counter parts (Dialogic, Synway etc.) SS7, ISDN, R2MF, SIP, Speech Recognition etc.

How does it compare with traditional CTI hardware and Propritary IVR Software ?

Though it claims to be Open Source, low cost but some how I am not convinced it is an alternative of Dialogic, Synaway or other cards. Because at the end of the day, you need proper support for the hardware, plateform while you concentrate on running your business, not the IVR application.

But I am not the right person about giving any expert opinion as I am not using Asterisk. But I have started reading Asterisk and its usefulness in developing and deploying IVR Applications.

I will keep updating about Asterisk here.