rob's blog

After an extensive consulting engagement with our social media marketing guru, we have decided to make an important switch to Cloud Computing. Watch this video to see how we arrived at this important platform choice for OnSIP.

Enjoy!

Actually, we always did.

Is this really a big deal? At OnSIP, it’s a split decision.

If you ask our CTO, he’ll say, ‘What’s the big deal?’ We’re just allowing the phones to do what they are made to do. If you ask our marketing folks, they’ll tell you “Fantastic, HD voice is really cutting edge stuff for a small business at a groundbreaking low price!”

Here’s the deal. Phones from Polycom, Cisco, Linksys, SNOM, Aastra and countless others currently come with support for HD voice, meaning they support calls in HD using some flavor of the G.722 codec, the “wideband codec”. For two people to have a conversation in HD, other than both having HD-capable phones, the only requirement is for the provider completing the call to allow two endpoints to speak to one another in the native G.722 codec. A provider would literally have to get in the way of the phones, preventing them from doing what they are inherently capable of doing in order to prevent an HD call. That’s why our CTO and others here say “big deal” to countless press releases from providers boasting about an all HD VoIP Network.

Because of our strict adherence to the SIP standard, OnSIP has supported HD calls since the first day HD-capable phones were available. But, as our marketing team will tell you, it’s been a long time coming that we are boasting about it.

Why? Our marketing folks felt we should boast about it when it truly was groundbreaking. And now, with HD conferencing, we really are confident that our support for HD is truly groundbreaking. We are confident you will find no other provider offering the following:

HD calls between:

  • Extensions – Inside your OnSIP account
  • Customers – Outside of your OnSIP account to any other OnSIP user
  • Internet Callers – Outside of OnSIP to any other SIP address
  • Conference Bridge Callers – Any caller to an OnSIP bridge, using an HD-capable phone and SIP address

Using:

  • Any HD-capable SIP endpoint (phone or software phone)

Yes, there are providers out there supporting the individual pieces, such as HD conferencing or extension calls using only a select few Aastra phone, but OnSIP is the first to support all of these features on any HD-phone as part of a complete business phone system at under $20 per user.

Should you care? Well, HD calls really do sound much clearer than anything on the PSTN. It’s a huge upgrade and we recommend our customers buy HD-capable phones to take advantage of it.

BTW, we also support ultra-wideband voice, HD video…..on any device that supports it. :-)

Hosted Unified Communications is hot! Or so they say. But what it actually means is up for debate. We have our opinion of what it is and we will get to sharing it very publicly later this summer.

In the meantime, there's lots of media covering the topic as a result of some research recently done by a few firms. Here are some stats...

According to Khali Henderson in a Phone+ article, Wainhouse Research puts annual revenue for Hosted UC at $200 million today and $5 billion in 2014.

Henderson states that "UC, at a base level must include three elements - telephony, messaging and presence - integrated into a single interface".

Does anyone offer that?

Do you know of provider who claims they do that?

Are you using a HOSTED service that does that?

We want to know.

Tell us what you think and share your experience. Please post a comment on our blog and let us know.

There’s lots of hype today about High Definition or Wideband Audio on VoIP calls. What does it all mean for business voip?

It’s actually quite simple. It means the sound quality on calls which are in HD, sound better than traditional phone calls. These days, I am used to HD quality calls but to be honest, my first experience on an HD call was shocking.

The difference between HD and “regular” calls is kind of like the difference between AM and FM radio. It’s substantial.

What makes it all possible? A VoIP call between two endpoints or phones is simply a session in which data is encoded and decoded according to a particular shared audio format. In the case of a High Definition or Wideband call, the shared format is referred to as a wideband CODEC.

OnSIP supports the G.722 ITU standard wideband codec. This is a widely used wideband codec so many phones support it, allowing OnSIP customers to have really high quality wideband calls to and from one another.

What’s really great is we have successfully tested wideband between phones of different manufactures including Polycom and Aastra. Around here, that’s very important as we support many clients with mixed phone environments.

In a business voip installation, wideband has a pretty big impact. Since many of the calls are between users (extension to extension calls), lots of calls are transmitted using the wideband codec.

Here are some resources about the use of wideband audio for business voip:

Using Polycom phones which support HD
http://www.wirevolution.com/2009/04/05/hd-voice-cookbook/

Making use of Wideband Codec Right Now
http://www.mgraves.org/voip/2009/07/making-use-of-wideband-voice-right-n...

Voip-info.org on Wideband VoIP
http://www.voip-info.org/wiki/view/Wideband+VoIP

Some providers make a big deal of their support for HD or Wideband Audio. At OnSIP, it’s just standard stuff.

I was flattered to see the CEO of Packet8, Bryan, has taken the time to evaluate our service, opening two accounts and then doing a side-by-side price comparison on his blog. When we started our company 5 years ago, Packet8 was a pioneer in our industry, setting a benchmark for business VoIP service. I can only imagine Bryan is a very busy CEO of a public company with hundreds of employees. I am honored that he was able to take the time to thoroughly evaluate our service.

I appreciate his thorough attempt to clear up any confusion about our pricing details. However, in the same spirit as his post, to clear up any factual errors, I feel it is necessary to present the correct pricing for OnSIP. Bryan’s analysis assumes a single user purchases an OnSIP package intended for at least 5 users, which is wrong. More appropriately, the correct analysis below assumes a company of 10 employees, not a company of 1 employee with an appropriate OnSIP package.

I personally have not tested Packet8 nor have I done an in depth analysis of the pricing for their hosted pbx service. I present Bryan’s prices for Packet8 from his blog post on Friday, December 5, 2008. His prices are based on the recently announced “December Sale”. The sale price requires a purchase of phones from Packet8 and an annual commitment. The regular price is 2x the sale price at $49.99 per user.

As Andy Abramson pointed out on VoIP Watch, OnSIP’s Always Available Pricing is still cheaper than Packet8’s half price sale by about 35% in the first year including startup costs:

OnSIP is cheaper than Packet8

In year two, OnSIP is still cheaper by about 30% even if the half price sale continues:

OnSIP is still cheaper than Packet8

Bryan’s analysis misses the key point: We do not have a per user charge. We only charge for what our customers use.

Bryan also stated that we recommend a “low-end” Polycom 301 phone and we sell an Aastra 53i phone. Lots of false statements here. We don’t sell phones. We don’t require you use any particular phone, only ones that support SIP, the industry standard on which Polycom, Cisco, Linksys, Aastra, and other leading vendors have committed to. My understanding is Packet8’s delivery platform is proprietary and standards-based SIP phones will not work with their service.

We do not recommend “low-end” phones. In fact, we attempt to dissuade customers from purchasing low-end phones as we conduct extensive testing and want to ensure a positive user experience for every customer.

Polycom phones are some of the highest quality IP phones available. About half of our customers use them and are extremely satisfied. And these customers self selected the phones; we don’t sell them at all. The Junction Networks team members use Polycom phones in our offices and our home offices.

Bryan does point out that our service does not include emergency services. This is a very important feature for our customers and we are currently testing our solution, which will be made available to customers in Q2, 2009. As with any other feature we deploy, we diligently work as hard as possible to ensure that when a feature is live, our customers can rely on it to work without exception.

Bryan also points out that everything we do is pre-billed. This will be changing shortly in response to some customer requests. We look forward to delivering on their requests and continued service.

And, as if we haven’t said this enough times, our always available pricing includes a free monthly trial, no commitments of any kind, no contracts, etc. You don’t need to talk to anyone to get any special pricing or fees waived from us. It’s always available and it’s always low. You don’t need to buy phones from us to get special pricing. You can use any standards based IP phone.

I hope this clears up any confusion about our pricing.

First, let’s address SIP: Session Initiation Protocol is the de facto standard protocol for establishing, conducting and ending a VoIP call. If you are really interested in technical details, check the Wikipedia entry on SIP.

SIP is to VoIP as SMTP is to E-mail. Just like the standard protocol, SMTP, which allows two email servers to exchange email data, SIP allows two endpoints (IP Phones) to connect to one another using a standard protocol. Without such a standard, two phones would not share a common set of instructions and guidelines in order to properly exchange packetized voice.

The great thing about SIP is that it promises to help do away with phone numbers all together. How? Imagine your email address could be “dialed” from a phone and it would kick off a call. Further, imagine everyone at your company is reachable by phone using email addresses. My name is Robert. It would be great if people could reach me by dialing Robert@xxxxxx rather than some random collection of digits, which have no meaning.

Think of SIP as the protocol for yet another service capability of a domain:

Email: SMTP
Web: HTTP
Voice: SIP
Video: SIP

OnSIP is a SIP service provider. With an OnSIP account, you can open your domain to SIP traffic. This will allow you and your team members to be reached by phone using email address too!

You can make and receive SIP calls to/from any SIP address. Calls to users on our network, since they are all SIP calls, are all free.

Check out our SIP Domain Features for more info.

Many people ask me what a virtual phone number is. Well, everyone knows what a phone number is. So the real question is “What is Virtual about a Virtual Phone Number?”

First, here is a dramatically oversimplified history lesson:

Historically, phone numbers were tied to physical locations. The phone company would provision a phone number to work over a single physical line, which would be “dropped” at the actual location the number would be tied to. Calls to that number could only be delivered to that physical location and businesses would have to receive the calls using expensive PBX systems which maintained routing smarts, voicemail applications, IVRs, etc.

With a virtual phone number, the physical limitation is removed, allowing a company to use a phone number in a more flexible manner with no reliance on physical presence of phone lines or phone systems. Calls to a virtual phone number are handled by a remote agent or proxy, which forwards on calls based on user defined rules. This allows a business to:

• Seamlessly connect multiple locations
• Eliminate on-premises telco equipment, telco space, phone lines, etc.
• Maintain phone service during incidents effecting physical offices.

Here is an example of how a business uses a virtual phone number from OnSIP:

Company X maintains a New York headquarters and a Los Angeles sales office. The company has local phone numbers and one toll free number, all virtual phone numbers. When a customer calls the toll free number, rather than having it answered by a phone system in either the Los Angeles or New York office, an IVR answers the call on the OnSIP Virtual PBX service. When prompted by the IVR, the caller selects option 2, for sales. Because there are sales associates in both offices, phones ring simultaneously in both offices until answered in Los Angeles. The call is from a key customer who needs to speak to the CEO who is working from his beach house in Cape Cod. The sales associate transfers the call to the CEO who is connected to OnSIP using his home office cable Internet connection. When the call is completed, the CEO uses 4-digit dialing to a make a free call to the sales associate in Los Angeles to congratulate her on a job well done for helping close a major sale.

The entire team is connected via OnSIP, which acts on behalf of the users, no matter where they are now or where they move. Users have the flexibility to make and receive calls and use the service as if they were in the office at all times.

OnSIP has phone numbers available throughout the country and are available for immediate activation.

Update - Here is some more helpful info about some industry terms which are often tossed around interchangeably:

hosted pbx

virtual pbx

business voip

I just read an article in the NY Times that purports to advise SMBs on how to take advantage of VoIP (Voice over Internet Protocol). While the story’s gist is true -- that VoIP calling in and out of the PSTN may save the SMB money – it dismally fails to paint the full picture of business telephony options.

Start with their first SMB recommendation:

“First off, if you don't already have a private telephone exchange (PBX), you'll need to install one. A PBX is a piece of equipment that switches calls between enterprise users, allowing a group of people (at a company or campus, for example) to share a specific number of external phone lines, saving the added cost of having an external phone line for each user.”

When was this written? These days, an SMB has no business buying, installing and managing PBX hardware. Today a huge number of SMB options include no hardware, eliminating the need for technical staff or consultants (aka ridiculously expensive support contracts) for ongoing PBX management.

Here is the beauty of IP-based telephony: There are no differences between PBX applications that run in your office on purchased or leased PBX hardware and PBX applications that are hosted in a data center, accessed over the Internet.

Oh wait -- except for these:

Hosted PBX services usually feature:

24x7 network operations management by specialized staff
Power, network and IP redundancy for fault tolerance
Disaster recovery options
No capital cost
Free upgrades
No need for support contracts
Scaling on demand as your business telecom needs grow or shrink.

Hardware PBXs come with:

Few or no redundancy safeguards
High capital costs
Ongoing maintenance requirements and costs
Specialized technical staff for management
Costly upgrades

Hosted PBX services usually offer SMBs no-penalty, low or no-risk evaluation periods. Many, such as our onSIP hosted service, even offer free trials. When evaluating a hosted service, customers can simply plug free or inexpensive IP phones into the LAN. In minutes, they can configure the phones to take advantage of the full feature set of a PBX, without any PBX in sight. A far cry from the vendor selection process associated with a purchased, customer-premise PBX.

There is little to no management of a hosted service on an ongoing basis. Moves, adds and changes can be handled in minutes by anyone who knows how to use a computer. Upgrades are free and happen without notice, so your service is never outdated and remains current with the latest IP telephony features and capabilities.

While the PBX hardware resides safely off-premise, total management control remains in the hands of the hosted PBX service customer, if so desired. Powerful yet user-friendly, web-based administrative portals extend total control of the service as if it ran on an on-premise PBX. Services also can be configured to support road warriors who want calls forwarded to cell or home phones, to route calls based on time of day, to create groups, auto-attendants and more.

The Times story takes pains to point out that the PSTN maintains “five 9s” of reliability, bettering a VoIP service. But does this consider real-world conditions? Does it take into account, for example, the excellent
chances of someone tripping over wires coming out of PBX hardware sitting under someone’s desk?

Today, hosted PBX services have more to do with rescuing SMBs from disaster than causing them. If a small business is struck by fire, flood, earthquake or just a crippling snowstorm, its hosted PBX platform can go on merrily receiving its business calls and forward them to employees sitting at any other spot on earth.

The same circumstances will crush, melt, or fill up the voice mail boxes of the PBX sitting in the telecom closet, VoIP or otherwise. For all these and other reasons,
SMBs these days should strongly consider the savings and benefits of hosted solutions before buying and installing their own PBX hardware.

While I wish I were considered as relevant as Bill Gates, at least at this point in time, we share a common vision of the end of the PBX. Microsoft recently unveiled their unified communications offering. At the launch in San Francisco, Gates stated "The transformation to software-based communications is going to be as profound as the shift from the typewriter to word-processing software." Hey, we feel the same way.

Wikipedia defines a PBX or Private Branch Exchange, as " a telephone exchange that serves a particular business or office, as opposed to one that a common carrier or telephone company operates for many businesses or for the general public."

We agree that large hunks of hardware, sitting in telco closets, serving individual silos or organizations are a thing of the past. We also agree that the future of unified communications is software based. We are headed in different directions though on our implementation of the PBX replacement.

Microsoft's unified communications services are embedded in existing Microsoft products that generally need to be managed in silos on a company by company basis. For some, this model makes sense. Junction Networks, on the other hand, delivers unified communications services as a service, with no managed software on the client side.

The funny thing about delivering unified communications services as a service is we are still calling it "Hosted PBX" for marketing purposes because that's what customers understand. The fact of the matter is the Junction Networks hosted infrastructure is so far from anything that acts or looks like a PBX, it is almost painful for us to call it a PBX. If anything, it is much like any web infrastructure, with routers, switches, web servers and db servers.

This model is far more scalable, redundant, reliable and cheaper than any per customer or per server model that was typically the model of the traditional PBX. Much like web or email hosting, the future of unified communications holds two options for a business:

DO IT YOURSELF:
Buy, install, manage and deploy software, such as Microsoft's products, with your own staff and resources.

COMMUNICATIONS AS A SERVICE:
Engage a service provider, such as Junction Networks, for a pay-as-you go service with little or no upfront investment or internal resource needs.

Both are valid models and which way a business goes depends on a particular organization's appetite for control, skill set of technology folks, and capital availability to name a few factors.

Happy to be on the same (or similar) page as Bill!

It seems Digium is no longer interested in taking a back seat to the some of the IP PBX vendors it helped spawn with the development of Asterisk, the open source pbx. In the last week, Digium first announced the purchase of Switchvox, the makers of an Asterisk based PBX. Only a few days later, Digium inked a deal with 3Com, who will distribute the Asterisk Appliance under the 3Com name. Both deals signal a strong move by Digium to take the lead in the SMB IP PBX race.

Digium says it reviewed a number of competing IP PBX vendors before selecting Switchvox. Speaking from experience, they made the right move. Junction Networks has partnered with Switchvox to provide easy-to-configure SIP and IAX Trunking since 2005. Our shared customers report the product is reliable, easy to manage and supported by a great team. It’s nice to see a great group of people and a great product get the recognition they deserve. It has been a pleasure for our team to work with Swithvox.

The acquisition pairs the Asterisk code makers with a superior GUI and high quality management team in Switchvox. Right now, the Asterisk based IP PBX race comes down to Asterisk vs. Fonality. If I had to make my bet, I would go with the guys who control the Asterisk code!

And because Digium controls the Asterisk code, big deals such as the partnership with 3com are obviously now within their reach. It will be tough for Fonality and others to make such moves to compete with the gorilla that is now Digium.

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