Hosted IVR services are in still nascent state as such, except Bulk Voice Calling, which has grown on the back of SMS ( Short Messaging Service) mass messaging after Mobile Phone revolution in India! But still, hosted IVR Service providers face many problems, majorly from telecom service providers.

Hosted IVR Service : Handicapped by Telcos

Lately, many companies have understood benefit of Hosted IVR services, but still they find the cost high. Though the cost seems much lower compared to IVR hosting Services in other countries like United States of America, United Kingdom etc.

Here are few problems which are faced by Hosted IVR Service providers in India :-

1. High Outbound Call cost

Though the mobile phone call rate in India is very cheap compared to other countries, but the story is not that simple for enterprise PRI line call costs. First, India is divided into 23 circles and there is differential charging for inter circle and intra-circle calls. Intra-circle cost is cheaper than Inter Circle ( known as STD or NLD ) cost. Then again, the cost for mobile phones does not apply to PRI lines. The cost for all circles uniform cost which is a requirement for Hosted IVR Service, starts with one rupee depending on the call volume committed!
But a normal customer is used to hearing about cost something like 1paise per 10 seconds which are available for Mobile Phone subscriber! So, it is really difficult to convince customers about high cost of outbound calls.

2. High Inter-operator Connectivity cost

There is a cost which customer has to pay is for inter operator connectivity cost! While mobile operators charge and collect payment from their subscribers for inter-operator connectivity, operators often go into dispute among themselves resulting in suspension of routing calls between them. Recently, both Reliance infocomm and stat run telecom company BSNL ( Bharat Sanchar Nigam Limited) had suspended connectivity between each other!

And it was only for enterprise solutions, not for normal Mobile of Telephone users. So it resulted many inconvenience to Hosted IVR service providers, Toll free number owners etc.

3. Frequent Downtime

Though India boasts of fastest growing telecom market, the quality of service is relatively poor! Though many telecom operators claim to have up-time more than 99.8%, in reality, it is much lower at 94-95% only! Since Hosted IVR service providers depend on Telephony infrastructure, they also suffer downtime accordingly which reduces customer trust on Hosted IVR services.

4. Problem in quick capacity enhancement

Many telecom providers provide PRI links from their BTS or switch using RF link. This has limitation of increasing capacity when required. So if some one wants to increase capacity, it takes some two months time to get new PRI connections. So, hosted IVR service provider has to either invest heavily and carry the burden of unused PRI lines, or wait 2 months period for capacity enhancements.

5. Lack of IP telephony

Lack of IP telephony in India is one hindrance, I think, which is hampering growth of IVR service, specially in call centre field. IP Telephony can not be bridged to any land-line or mobile phone network due to security reasons. With high outbound calls and lack of IP Telephony is hampering many hosted IVR applications being provided by IVR hosting companies.

VoIP or IP Telephony can lower high outbound cost in India.

6. Tier 2 & 3 cities do not have many Operators

Smaller cities, apart from metros, not many telecom companies have built up infrastructure to provide PRI lines. Presently, only BSNL and Reliance seems to be capable of provide PRI connectivity in all cities in India. This has also resulted in lack of competition and monopoly as well as poor service and own time.

So, till the telecom infrastructure and quality of service by telecom service providers, Hosted IVR service in India will always be handicapped!

*With feedback from Ankit Bhargava, Director – Marketing, Niche Tech Services (P) Ltd.

Donjin’s Keygoe series of multimedia switches are communication platform which can handle almost all types of voice, video applications in any size. While the entry level KeyGoe Series product Keygoe1000 starts with 1 channel and can be increased to 240 channels, and high end KeyGoe Series like KeyGoe 3000 has up to 127 E1 lines.

KeyGoe Donjin Multimedia Switch

All KeyGoe Series products support all types of available telecom protocols like ISDN PRI, SS7, IP ( RTP, H.323 etc). So, a KeyGoe series of product is almost future proof which will never go obsolete. Keygoe series multimedia switches are independent rack 19″ rack-mountable.

In their voice resources, they support all types of CODEC, play audio files of many formats like WAV, ADCPM, Dialogic VOX format. They have inbuilt echo-cancellation, AGC ( automatic gain control), full duplex audio playing and recording. They have VAD ( Voice Activity Detection) as well as CNG ( Comfort Noise Generation).

KeyGoe series products provide 1:1 conference facility for all voice resources. They have built in FAX resources too.

They support Windows Operating System, Linux ( Kernel Version 2.6 and above) and have SDK with many sample codes in C++, C#, Delphi.

The installation and configuration is also quite easy with visual tools to configure various parameters.

KeyGoe Series products are very feature rich, low cost but performs very well. All types of IVR applications as well as telecom solutions can be provided using KeyGoe series of products. Developing Voice applications, Conference application, FAX applications are quite easy using Donjin KeyGoe products.

Donjin support has improved a lot over the time and they have now a big number of installation in India. KeyGoe series products are being used in Hosted IVR solutions, Voice based Value Added Services, Hosted Conference Services as well as hosted Outbound IVR ( Bulk Voice Calling).

KeyGoe series products from Donjin are available at Sunshine Telesoft (P) Limited. in India.

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!

IVR Hosting is growing world-wide and many people are increasingly opting for Hosted IVR solutions. Though hosted IVR Solution may not be able to replace On-Preimse IVR solutions in some cases, but here are top five reasons why one should opt for IVR hosting :-

IVR Hosting ay be your ideal IVR solution!

1. Future Proof

With increasing number of CTI vendors as well as available CTI boards, it has not only become difficult to choose a CTI board, but it has also created a new risk, that is obsolescence! Many people may remember buying good old D/4PCI without any fear or consideration of going obsolete, but not anymore!
With recent advancement in speech recognition technologies, as well as complex IVR systems which may require data exchange between servers located at many places, one can safely say that, today’s IVR technology is going to be obsolete a lot quicker than before.

In hosted IVR solution, this is never going to be a problem for user! One can expect, the hosted IVR service provider will always be updated with latest technologies, features available in IVR technology.

2. No maintenance and no support nightmare

IVR System consists of more than one components. For example, it has telecom components which connects you to the PSTN ( Public Switch Network). Then there is a computer Server and power as well as power back up. Apart from this, there is one CTI vendor as well as IVR Software vendor!

Most of the times, you do not find one single vendor providing you all above components. So, one needs to contact all these vendors for support. One may even encounter a problem like, CTI vendor blaming the Telecom connectivity for the IVR system malfunctioning, while the computer hardware guy blaming the IVR software.

In hosted IVR solution, you have only one problem, only one support, either your IVR is working or it is not. And contact only one support for this. It will save time and resources for any company.

3. Real-time monitoring of expenses and exit route

With IVR hosting service, one can actually measure ROI ( return on Investment) quickly and in real time. From the monthly payment to the Hosted IVR service provider, one can measure exact number of calls, call abandonment, successful calls and any saving in human intervention. But one may argue, these data are available on On-Premise solution too. Yes, but when you know use of IVR is not reducing burden of human resources, you have had already invested on the IVR systems as capital expenditure!

But with Hosted IVR solution, you have quick exit route or try out different type of call flow, different features. On premise IVR, one can not expect these benefits!

4. Focus your core business, no allocate resources for IVR a business tool for you

IVR system is a business automation system to automate customer care, reduce human intervention, increase accuracy of information as well as collecting measurable information about callers and customers. Maintaining or developing IVR system may not be your core competence. So, one should devote resources in their core competence rather than devoting resources in IVR upkeep.

Hosted IVR is the perfect solution which offers complete IVR solution without having to know anything about IVR yourself.

5. Quick upgrade and expansion

Hosted IVR offers facility to upgrade capacity of calls, or any other required feature quickly. One can also opt for using different technology for different kind of calls. For example, one may opt for VOIP for outbound calls while using TDM network for incoming calls by the customers. The best part is, one can change this kind for arrangement frequently as per requirement. With On-Premise IVR, one is stuck with the CTI board purchased as well as initial SRS ( System Requirement Specification) of IVR.
These kind of flexible features are available with IVR hosting and if one requires these facility, he might have invest a lot in different hardware alone!

So, it may be time for you to think about IVR hosting seriously!

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!

Dialogic D/300 JCT-E1 is the entry level board for anyone who needs digital network interface with PSTN network as well as voice resources in the single board. I feel, D/300 board is most used entry

Dialogic D/300 JCT-E1 board

Features of Dialogic D/300 JCT-E1 board

1. Digital Network Interface

It has both E1(2.048MBPS) and T1(1.544 Mbps) support. It supports digital interface network protocols like ISDN PRI, R2MF. ISDN PRI is widely used protocol for connecting CTI board with PBX systems. With D/300 board, it is quite easy to configure it to work with many ISDN variants as per PBX from one country to another or manufacturer.

2. FAX support

Dialogic D/300 board supports both FAX and host based speech recognition. In certain cases, FAX and speech recognition may not work together. It also supports CSP ( Continuous Speech Processing) which offers IVR developers a powerful feature to develop many interesting IVR applications and features depending on the call progress. Speech recognition feature provides IVR developers to develop Speech-Enabled menu for any IVR application.

3. Scalability

Dialogic D/300 supports H.100 connector and it allows it to connect to other Dialogic boards. Inexpensive normal PC can be used to install this board and one PC may install upto 16 boards per machine if PCI slots are available.
With its support for GSM and G.726 bit exact voice coders ( standard for Voice Profile for Internet Messaging (VPIM) , one may develop unified messaging solutions based on D/300 board.

4. Programming

Dialogic D/300 board is supported by many RAD Tools for IVR development. It also supports widely popular GC ( Global Call Protocol) for making many high end, mission critical IVR applications.

Applications

D/300 board can be used to develop any kind of IVR applications. It offers many call related information available with digital network interface protocols. Call bridging available in ISDN PRI is one of the feature which is used to make Click To Call application can be developed.

D/300 board is also ideal choice for starting a voice portal service to start with 30 lines and then increase the lines.

DIalogic has some other models with higher capacity with same features. D/600 and DMV1200 have 60 voice and 120 resources respectively. One may move from D/300 to DMV/1200 without having to make any changes in the application.

Other CTI vendors like Donjin, Synway, Digium too have D/300 equivalent boards. For example, Digium TE122P/TE122B, Synway TEJ Series and Donjin DN300 are Dialogic D/300 equivalent boards with similar features.

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.

VoiceXML is the open standard markup language for all kind of voice applications. Does it include IVR ? Yes, it does. VoiceXML is the modern and latest way of writing IVR Software, develop voice portals and much more!

“VoiceXML 1.0 was published by the VoiceXML Forum, a consortium of over 500 companies, in March 2000. The Forum then gave control of the standard to the World Wide Web Consortium (W3C), and now concentrates on conformance, education, and marketing. The W3C has just published VoiceXML 2.0 as a Candidate Recommendation. Products based on VoiceXML 2.0 are already widely available.” Source

How does one use voiceXML ?
VoiceXML is used the same way as HTML is used! So, voiceXML is the HTML of voice web! And even its architecture is same as that of HTML. And it is used for voice portal as in web portal!
1. It has a voice browser
2. It has a interpreter
3. And finally it has hardware implementation

Advantages of voiceXML
1. it is open source
2. It is telephony hardware independent
3. It can run along with web applications
4. Faster development of voice applications
5. There are now RAD ( Rapid Application Development) tools available for VoiceXML based IVR system development.

Disadvantages of voiceXML
1. Many sophisticated Telecom Protocol specific information like DNI etc information may not be available.
2. Many voice playback options are not available yet.
3. VoiceXML may not be suitable for large and complicated IVR system.

While I am still studying about voiceXML, I would update the post in future soon.

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.