Enjoy asking and answering discussion questions on TelecomYou, and enter the Raffle to win $200 USD. It is amazing … the accidental but awesome business development that will result, in the meantime.

Discussions section link: http://telecomyou.com/forum/

Other than the raffle, the most active Discussions member will get the “Most Active Forum Member” award for the month of April, 2011. The Award badge will appear at the member’s public profile page.

The more questions you enjoy asking or answering; the higher your chance to win the prize.

Some things that people are talking about now:

Which tablet do you recommend?

What is your average work day like?

What should every newcomer know about the telecom industry?

Has anyone tried iPad 2?

Where exactly is Telecom Business is going?

Feel free to start a new question; this will help others to learn, teach, and collaborate. Tell them Techistan sent you! Visit http://www.telecomyou.com/forum now. Sign up if not already and enjoy!

Plus, looking for a job in Telecom. Watch and listen to Sheraz show and explain how on Telecomyou.com.

TelecomYou is the place where international and local IP communications movers and shakers like those on DIDXchange congregate, innovate and collaborate with real lucrative business and technical development.

Don’t miss out. Sign up now at http://www.telecomyou.com. This video explains how to add friends from your network sites such as MSN, MySpace, Gather, Facebook, Jumo, Linkedin,  TelecomYou etc. It was created by Alishba Joseph of Super Technologies, Inc.

And other Techistan TelecomYou Tech Talks over there!

TelecomYou is the place where international and local IP communications movers and shakers like those on DIDXchange congregate, innovate and collaborate with real lucrative business and technical development.

Don’t miss out. Sign up now at http://www.telecomyou.com. This video explains how to Update status on TelecomYou. It was created by Alishba Joseph of Super Technologies, Inc.

And other Techistan TelecomYou Tech Talks over there!

Share a bed time story with your children. Nice way to share technology information and maybe an infomercial here and there, tiny ones, that is!

Don’t worry. Techistan’s iPad is on the way for testing. We’ll write a review after a thorough examination and ultrasound.

WHITE PLAINS, N.Y., Jun 24, 2010 (BUSINESS WIRE) — Vox Communications, a provider of Voice over Internet Protocol (VoIP) and smart phone applications, and a wholly owned subsidiary of Pervasip Corp. (PVSP 0.07, -0.02, -17.65%), announced that it has released a downloadable application for mobile VoIP for use on the Apple iPad, iPhone and iPod mobile devices. Consumers are now making telephone calls on the Vox network from their iPad, iPhone and iPod touch devices.

Pervasip’s Chief Information Officer, Mark Richards, noted, “We continue to further our reach to more mobile and portable devices at affordable rates with our VoIP application. The VoX SIP credentials are easily delivered to Apple’s mobile devices by downloading the fring social network interface from iTunes. VoX is commencing social network media marketing to an addressable market of over 80% of the installed smart phone base, or more than 100 million devices. Our mobile VoIP plans start at $4.95, which is a low cost of entry to try our quality and reliable VoIP service offering. VoX also offers 500 minutes for $9.95, which is less expensive than Skype, and $29.95 for unlimited calling to the US, Canada and Puerto Rico. We have also released our reworked ordering web portal to allow the entry of non US based addresses so that we can now target a global audience looking to make mobile calls to the US, Canada and Puerto Rico from any country in the world, over Wi-Fi or a 3G data connection.”

About Pervasip

Pervasip Corp. delivers wholesale voice over IP (VoIP) telephone services for the residential and small business markets through its wholly owned subsidiary, VoX Communications. Leveraging a nationwide VoIP network and internally developed proprietary software and product features, VoX offers high quality voice services, flexible back-office capabilities and automated provisioning systems. Vox recently entered the mobile VoIP services and applications arena, which is expected to approach 300 Million users by 2013. It offers a feature-rich, low-cost, high-quality alternative to traditional phone services. For more information, please visit www.voxcorp.net.

Forward-looking statements: This release contains forward-looking statements that involve risks and uncertainties. Pervasip’s actual results may differ materially from the results discussed in the forward-looking statements. Factors that might cause such a difference include, among others, certain risks and uncertainties over which the company may have no control. For further discussion of such risks and uncertainties, which could cause actual results to differ from those contained in the forward-looking statements, see the discussions contained in Pervasip’s Annual Report on Form 10-K for the year ended November 30, 2009 and any subsequent SEC filings.

SOURCE: Pervasip Corp.

At Pervasip:
Paul H. Riss, 212-404-7633
Chief Executive Officer
phriss@pervasip.com
or
Investor Relations:
Brian Holden, 913-226-3818

Copyright Business Wire 2010

Listen for a podcast interview with Pervasip’s Vox Communications.

What’s up in telecommunications in May 2011? One rather new powerhouse ITW and an even newer powerhouse from TIA. In fact, the TIA 2011 debut is May 18-20, 2010 in Gaylord (Dallas) Texan, Grapevine, TX, and International Telecoms Week in Washington DC is scheduled May 24-26, 2010.

The Telecommunications Industry Association is pleased to announce the debut of TIA 2011: Inside the Network, May 18-20, at the Gaylord (Dallas) Texan, in Grapevine, TX. This will be the first event of its kind for the telecommunications and ICT industries, for a number of reasons:
1. A focus on the challenges and opportunities relati

VOIP Today magazine (www.voiptoday.org) is privileged to conduct exclusive interview with Kamailio (OpenSER) co-founder and core developer of OpenSER and SER projects Daniel-Constantin Mierla. (Read on and head over to iTunes or DIDX podcasts to listen to Techistan’s Suzanne Bowen interview Mierla in 2009.)

Kamailio (former OpenSER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second. Among features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video), SIMPLE instant messaging and presence, ENUM, least cost routing, load balancing, routing fail-over, accounting, authentication and authorization against MySQL, Postgres, Oracle, Radius, LDAP, XMLRPC control interface, SNMP monitoring. It can be used to build large VoIP servicing platforms or to scale up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS.

VOIP Today magazine spoke with Daniel-Constantin Mierla about Business , technology and products .

voiptoday.org:   What is different about Kamailio and other open source telephony solutions ?

 Daniel-Constantin Mierla: Kamailio is a SIP server, not a pure telephony engine like Asterisk or FreeSWITCH. What means that? Kamailio handles only the SIP signaling traffic, no media (e.g., audio or video) processing, therefore using Kamailio for telephony is just a subset of what can do. Of course, one of the typical use cases is for telephony, but as easy as that you can use Kamailio for instant messaging and presence or machine to machine communication.

Because of its design and purpose, Kamailio targets performance and stability in first places. When we talk about performances we mean more than 5000 call setups per second and more than 300 000 SIP phones registered to same SIP server instance. These figures show also why Kamailio is used very often as load balancer in front of IP PBXes, PSTN gateways or media servers.

voiptoday.org:    What is your average annual growth rate?

Daniel-Constantin Mierla: I could see interesting trends during all these years.

First, there are not many options out there for this kind of needs and all are based on the same initial project SIP Express Router (SER). Since 2008 Kamailio (which started as fork of SER in 2005 under the name OpenSER) and SER teams work again together, so we practically talk about same application, yet different names.

Back to your question, the project started in a research institute, Fraunhofer Fokus, Berlin, in the very early days of SIP and VoIP. After SER was released as open source in 2002, we had a good rate of adoption in academic institutes.

Next wave was realtime communication innovators – they couldn’t afford to wait big vendors implement their needs, therefore this was their chance.

Today is clear that SIP is the protocol for telephony services. More and more of the classic telephony companies started using it. I can say also that building SIP trunking services is very popular.

 To understand where we stand today as a project, here are some numbers:
•    since its beginning, the project counted over 70 registered developers and hundreds of contributors
•    the users mailing lists have over 63 000 posts
•    the development mailing lists have over 40 000 posts
•    over 2000 commits during 2009
•    over 170 extension modules
•    over 3000 downloads per month only from main web site
•    project management: 11 individuals (from 5 countries)

voiptoday.org:    What were your greatest achievements and implementations since your beginning on 2001?

Daniel-Constantin Mierla: It is hard to count the implementations, I was directly involved in several hundreds so far. In the first stage of the project, the biggest achievement was making it the reference implementation for a SIP server and keep it so.

One of the largest deployments I am aware of is 1&1 Germany – they have about 4 millions phone lines configured, routing more than 1.5 billions minutes per month. These are public data anyone can find on the net, but there are other big deployments and the companies keep them confidentials. Anyone can go to project web site and look at the references page for more examples.

Kamailio received the Best of Open Source Networking Software award at BOSSIE’09 by InfoWorld magazine, a recent project achievement that make us very proud of and confirmed us its spread and popularity.

voiptoday.org:    What are the future plans for Kamailio, both short term and long term?

Daniel-Constantin Mierla: In short term, there is a lot of effort in improving asynchronous processing to face the larger signaling traffic demands brought by instant messaging and presence. Right now the missing part is asynchronous TLS, we have asynchronous SIP message processing and asynchronous TCP since release 3.0.0. TLS should be added in a matter of few weeks and released with 3.1.0. You cannot find these anywhere else.

Another short time goal is to make the maintenance straightforward and adoption easier for new comers. We talk here about a SIP routing server, that means an administrator needs to know pretty much about the protocol itself. People managing telephony systems or IP PBXes don’t interact usually with protocol layer, they deal with calls, not much importance for them if beneath is TDM, H323 or SIP. Here we work to make their life easier.

For example, in the upcoming release we added interactive configuration file debugger, a nice way to troubleshoot your config as well as understand what is going there. Also, we added include and define directives that allow to separate the SIP routing logic from parameters, so system administrators have to change just few lines to adapt the config to their needs. Asipto released SIREMIS (http://siremis.asipto.com), a complete web management interface that helps to do provisioning easier.

voiptoday.org:    What is the competitive advantage for Kamailio?

Daniel-Constantin Mierla: Reliability was always the strongest point of this project. Configure it properly, start it and forget about it – simple as that. Core team is more or less the same since 2002, one left out of four, but others joined and there are over 50 registered developers, about 20 being very active. SIP v2.0 was released in 2002, this shows the expertise the team carries on.

Processing capacity is also an important characteristics – with an usual server you buy now for less than $2000, one can handle several ten thousands of registered phones and thousands of active calls.

voiptoday.org:    What kinds of platforms like Skype and OCS do Kamailio support?

Daniel-Constantin Mierla: Haven’t tried myself, but I heard Skype offers SIP interconnect right now, that makes Kamailio – Skype communication straightforward. Integration with OCS for trunking is possible via TCP or TLS, using SIP as well.

Practically any device that speaks SIP just works with Kamailio. Over the time I combined it with Asterisk, FreeSWITCH and many PSTN gateways or Media servers such as Cisco, Audiocodes, Telcobridge, Teles, etc.

voiptoday.org:    What is kamailio next product and what distinguishes it?

Daniel-Constantin Mierla: Kamailio 3.1.0 is the next major release, scheduled for this autumn. Novelty is not missing at all. Some new features were mentioned already, like config file debugger or asynchronous TLS, the others include embedded Lua and Python on configuration file, interprocess message queues, geo IP API, registration to remote SIP server, tree-based caching container and several more listed at:

http://sip-router.org/wiki/features/new-in-devel

voiptoday.org:    Does Kamailio ‘s solutions support any open source codecs?

 Daniel-Constantin Mierla: Good question since it allows to clarify some things. The answer is short: yes, all that are available and those yet to be released. Why? Because we do not process media. If you build devices that speak SIP for signaling and they are able to transmit 3D video, Kamailio is ready for it. Understanding media stream is an end-to-end affair, we just connect origin with destination.
voiptoday.org:    What is the average cost for an integrated solution provided by Kamailio?

Daniel-Constantin Mierla: Kamailio is the open source SIP server, the application, like Asterisk. Then there are several companies involved in development and building solutions with it, like Digium and other companies for Asterisk. More or less the same, Kamailio has a business environment  around the project.

I can talk about my company, Asipto, solutions span from low range of load balancers to high end complete unified communication platforms, with a save of at least 30% comparing with telco vendors and the main benefit of owning everything and being able to enhance as you wish, the power of code is in your hands.

voiptoday.org:    How did the slow economy affect the business?

Daniel-Constantin Mierla: Global economic crises affected everyone, fortunately we were affected less that others. We play in cost effective solutions market and price became a major criterion to select a product. Furthermore, the reputation of reliability and the number of features keeps us in a good position on the market.

voiptoday.org:    What are your predictions regarding market expansion in the future?

Daniel-Constantin Mierla: We were constantly growing since the project started. We are ahead implementing features that would really get the momentum in very near future. Just want to mention here TLS for secure communication and spit protection, then IPv6 which gains more market every day. Both of them are very well supported right now, so we are ready for that moment.

voiptoday.org:    Do you carry on any special events or exhibitions?

Daniel-Constantin Mierla: We do have dedicated events. At least once a year we have developers’ meeting, next is going to be on 8th of June, in Berlin. It is face to face meeting, open for community members as well, were we discuss existing issues and brainstorm future development.

On business side, we used to tie with various events where we had dedicated tracks, running one or more days. Lately we went for smaller events, targeting local or country-level markets. In the business we act, massive call volume routing, you don’t have too many potential customers in a region. Trying to get personal is important. In this fashion we run this year The Presence and Future of SIP Routing, first was in London, UK. Soon we will announce dates for other places in Europe.

Besides those, we participate to many related events. SIPit 26 was las week in Sweden, where I participated to test, main interest for me being TLS and IPv6. Then we have presentations at Amoocon 2010 in Rostock and LinuxTag 2010 in Berlin in less than one month – at LinuxTag the project has a booth for the entire 4 days exhibition.

Another group of events are trainings and certifications, next to come are Miami, USA, June 21-23, and Malaga, Spain, July 5-9.

Techistan interviewed John Lutz, Mission Integrated Solutions of Qinetiq, at CTIA Wireless 2010 in Las Vegas. The popular topic of voice over Internet, as much as it saves money and is full of features and flexibility, some still hesitate to use it for mission critical situations.

Qinetiq is a products and solutions provider for the U.S. government, local and state governments. Communications is John’s focus. Typically, Mr. Lutz was at CTIA looking to enhance the Qinetiq product line. There were thousands at this conference searching for services and products worth investing in.

John Lutz was also attending conference sessions and listening to hundreds of conversations around him to understand areas that are currently underserved that his company could assist.

There are two issues John Lutz believes that are keeping people from taking advantage of or moving to voice over Internet. For small businesses it is that traditional alarm systems that many of them have require a landline service, and voice over Internet is not compatible. For residentials, the problem is the 911 provisions.

“So what about GPS? Google™ can find us. Why can’t 911 be mashed with that?” asked the interviewer.

John Lutz replied, “Personal privacy …”

Find out more about Qinetiq at http://www.QinetiQ-NA.com.

VoIP services for SMBs are projected to grow faster than any other VoIP service.  If you’re a competitive service provider, then you need to know how to profitably serve this fast-growing market and drive new revenue, maintain healthy margins and set enjoy long-term growth!  Target the market that the big service providers have ignored!

IPsmarx has recently published a whitepaper entitled, “Three Keys to Profitably Serving the SMB Market,” where you can find out how to capitalize on the SMB opportunity. 

Understanding the needs of the SMB market is critical.  But, the technology is what will differentiate you. You need a solution that offers all the services an SMB needs, with user-friendly online access and management tools to easily manage the service from customer order to customer invoice, and promotional tools to help you grow your customer base.  And you can do it while generating healthy operating margins.

An All-In-One integrated platform for easy and complete SMB services and management: call routing, billing, invoicing, reporting, and marketing, with a unique customer management console giving customers information and control and information of their services and payment options.

Promotional marketing tools to help grow your business with a customized, integrated E-store, promotional tools such as email alerts, and a wide variety of payment options, such as coupons and vouchers, to create compelling offers and convenient payment alternatives.

Unparalleled reliability with redundant servers, systems scalable up to 2000 concurrent calls on a single server at 100 calls per second, dedicated QA engineers, accounting testers and ethical hackers, all designed to help you lower operating costs, and obtain greater efficiencies.

Download this whitepaper “Three Keys to Profitably Serving the SMB Market,” now to find out how you can grow your business with new customers, expand your capabilities to existing customers, simplify operations and reduce costs, all with a highly reliable and scalable solution. 

Learn more at http://www.ipsmarx.com/marketreadysoftswitch/whitepaper.html or reply to me for more information.

Submitted by and for more information, contact:
Ludymar Reveron (Listen to a podcast with her on iTunes at http://itunes.apple.com/us/podcast/didx-podcasts/id322949120.)
Sales Manager
IPsmarx Technology Inc.

Rob Adler of Vantage Communications and Humberto Correa of FonicaPBX met at the DIDxchange booth at CTIA Wireless 2010 in Orlando, Florida. First topic was the 32 page petition by AT&T to project all copper lines to voice over Internet protocol within the next ten years. Verizon also petitioned to do the same within seven years.cc

Lorraine Baines: Wait a minute, don’t I know you from somewhere?

Three point seven million lines per year are lost by both Verizon and AT&T? To whom? VoIP-hosted companies?

Dr. Emmett Brown: There’s that word again; “heavy”. Why are things so heavy in the future? Is there a problem with the earth’s gravitational pull?

Rob Adler adds, “All these CLECs ride along the Verizon or AT&T copper lines. If AT&T and Verizon get out of the copper line business, these guys do not have anyone to maintain their wires because they are not IP.”

Milton Baines: What do you mean, you’ve seen this? It’s brand new.

Hosted IP vendors such as Fonica, JKL5Group, Brekeke, Comtel-Networks, Artisit PBX, VirtualPBX, and PortaOne are filling in the gap. Humberto declared that we are going “back to the future” to 1996.

Back to the Future quotes were humorously sprinkled in this article.

More information:
Verizon and AT&T challenge FCC role in broadband, too. http://www.washingtonpost.com/wp-dyn/content/article/2010/03/24/AR2010032403106.html?hpid=sec-tech
http://www.fonicapbx.com
http://www.pr-vantage.com/
http://www.xchangemag.com/articles/beyond-small-biz-clecs-turn-to-hosted-voip.html
http://www.techistan.com/2010/02/01/co-opetition-rules-hosted-voip-the-new-clecs/