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This article is 100% spot on. It's been impossible to hire an engineer (software, network, etc.) in NYC in the last 20 years. Wall Street gobbled them all up. Now we're bailing out Wall Street.

Let's get back to actually building things. We have smart people, let's put them to work making this country better, not just making money.

I was over on Freeswitch's website this morning and noticed a link to a list of 104 Cisco and Avaya alternatives for IP PBXes. I thought it was an interesting list. Even though it does not seek to evaluate the IP PBXes, it is pretty astounding to note just how many options are out there. The Asterisk team should be particularly proud, given how many names on the list are Asterisk derivatives.

Although there were zero doubts in this corner, the sheer number of choices out there is a testament to the importance of VoIP for the future of communication.

Here's a little trick to help make things easier as we find the holiday season upon us. Will your office be closed over the holidays? Would you like to keep your customers in the loop of your holiday schedule? Would you like calls to automatically roll over to voicemail when no one is scheduled to be in the office? You can do all of that with Junction Network's OnSIP Hosted PBX by using business hour rules and announcements.

To keep your customers in the loop on your holiday hours:

1. Create a new recording under 'Resources' or by dialing *94 from any OnSIP phone. Record your holiday hours message as a recording.

2. Create a new 'Announcement' under the 'Apps' tab. Name it something easy to remember like 'holiday'

3. Re-record your attendant menu greeting to say something like "Press 4 to hear our holiday hours"

4. Edit your attendant menu so that 'on the press of 4' the caller now hears your new Announcement created above.

To roll calls over to voicemail when no one will be in the office:

1.) Create a business hour rules for the week of 12/25 and 12/31. Business hour rules are free.

2.) Set it up so that when users will not be in the office, Tuesday and Wednesday for example, you set the business as 'closed'.

3.) Put them in place on Monday (or any time that week before the holiday) and let them run for the week.

You can create as many Business Hour rules as you need and apply them at any time, but it's a manual process. Business hour rules are day-of-week and hour-of-day based. Be sure to return to normal operation (normal recordings and normal business hour rules)upon your return.

We've been so busy lately here in the engineering department that it's difficult not to feel a little bit like an elf, what with the holiday lights and snow outside. We have some exciting new projects that we're hard at work on that will make being a Junction Networks customer even better, but nothing that's ready to announce quite yet.

In the meantime, we were mentioned in Voip News, in a nice article that covers the A to Zs of getting involved with a hosted VoIP provider. It's an excellent checklist for anyone considering the jump to a hosted VoIP PBX, like our OnSIP Hosted PBX.

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:

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

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.

SIP via UDP vs. TCP

One of the things I enjoy doing is taking complex subjects and explaining them in such a way that nearly any interested party can understand. I spent nearly an hour last night explaining the Big Bang theory to my 12 year-old son. I think, eventually, we got there. The task before me today is to explain why SIP, the protocol of VoIP, is best left as a UDP service as opposed to a TCP service. This is nearly as formidable as explaining the Big Bang to a sixth grader.

First, we need to build some vocabulary. TCP and UDP are connection protocols in use today for data traversing the Internet. Data travels across the Internet in packets. Think of them like letters. Like letters, they have an envelope with a to/from address on them. TCP and UDP are just two types of envelopes. The IP addresses are the to/from addresses. They both carry data and both use IP addresses, but just the outside envelope is different. Think US mail vs. FedEx. The address on the envelope is the IP address for where the packet came from (source address) and where it's going (destination address). TCP is so prevalent on the Internet that it's typically combined with IP, and written as TCP/IP.

TCP would be the 'FedEX' part of the analogy from above. Whenever two servers 'speak' TCP, they set up a formal connection. Every time a packet is sent from one side, the other side sends a packet back acknowledging the packet's arrival. If no acknowledgment packet arrives after a certain amount of time, or if the acknowledgment states that there was a problem, then the packet is re-sent. It can sometimes take a few seconds for a packet to be fully successfully transmitted. TCP is optimized for accurate delivery, not timeliness, and is the protocol for WWW sites and e-mail among others.

UDP is the opposite. It is a protocol optimized for getting packets there in a timely fashion with little overhead, but with just as little accountability. It's more like throwing a bottle in the ocean. Ok, a very rapidly moving ocean, but you get the point. With UDP, you just address the envelope and drop it on the network. There's no handshake, and no setup. Just the data packets. UDP is meant for real-time conversations where you are more interested in keeping the stream going then making sure that you have every single packet.

Right this very moment, Junction Networks has over 3,500 devices attempting to connect with Junction Networks. These devices are everything from individual SIP phones, to SIP devices, to other PBXs. Most of the connection attempts are simple SIP registrations. A SIP registration is when a SIP device tells the server, in this case Junction Networks, that it's available for calls and what its IP address is. This communication happens anywhere from every minute to every hour, for EVERY device. That's a LOT of packets. If these were TCP packets, each time a phone wanted to tell us that it's available, it would have to go through the whole TCP connection setup. That would be a HUGE amount of overhead for a VoIP carrier. In a LAN environment, it would be manageable, but for thousands of individual devices and hundreds of them attempting to register EVERY SECOND, a TCP connection would grind servers to a halt.

Next, once the phones are registered and a call is set up, it's really UDP's time to shine. A phone conversation is a stream of packets meant to be created, sent and received in real time. A lag, any lag, would mean a degradation in the quality of the phone call. Imagine hearing something on the call one to two seconds after the person on the other end says it. You're replying to what they're saying, but they've already moved on. It would be totally disconcerting. And, since it's real-time, there's no catching up. Better to drop a packet and have a millisecond of silence than seconds of lag.

In short, VoIP traffic is best left as UDP traffic for both server load and call quality reasons.

This post gives me the background I need to answer why BitTorrent will NOT bring down the Internet and VoIP if/when they switch to UDP, contrary to
this article. Taking on explaining BitTorrent is going to be much harder than the Big Bang.

Junction Networks is pleased to announce that we've been nominated as a finalist for Small Business Computing's Excellence in Technology Award in the VoIP category. The focus of the Excellence in Technology Award is to acknowledge companies that have assisted small business owners in running their own business more productively. If you'd like to vote for us (and we hope you do), you can do so by visiting Small Business Computing and clicking on the Choose the Winners link.

There are a number of products to vote on, with categories in hardware, software, security and e-commerce. I personally can't wait to see the results!

I was sad to hear today that Tanta, a blogger at the finance and economics blog Calculated Risk, passed away. For me, the world will always divided into pre-Tanta and post-Tanta. She will forever mark a change in my reading habits. Outside of my narrow professional field, traditional media and journalism had always made up my primary source of information when trying to make sense of current events in the world around me. But that was before I stumbling onto one of Tanta's posts which shown bright and clear against the murky and frustrating background of noise being spewed forth by the usual suspects. From that day forward, I've looked for and found that the best signal to noise ratio on just about any topic can be had in the blogosphere. Thank you Tanta. RIP Tanta.

Andy Abramson just posted an article discussing SIP vs the calling card. I found it particularly apropos, as I've just returned from a two week vacation in Europe. It's been a few years since I've been in Europe, but what I've always done in the past is get myself to a local newspaper store and purchase a calling card in order to call home and tell my friends what an awesome time I'm having without them. I tried to do this on this vacation, except that I discovered that calling cards seem to have become obsolete, at least in the stores of Aberdeen, Scotland.

However, what I did discover in the local chain supermarket was that I could buy a pay as you go (practically) disposable cell phone. In the supermarket! I could also buy minutes to charge my phone with, for not too much more (including the phone) than I would have previously have spent on a calling card. I was thrilled at the convenience.

Why wasn't OnSIP Hosted PBX the right answer in this scenario? This vacation happened to involve a lot of travel to farms and other remote locations where cell signal was spotty and an Internet connection was out of the question. So one of the downsides to buying a U.K. cell phone was that I had to pay international rates when I called home, which I wouldn't have had to do if I were using a SIP phone and my OnSIP account, but on the other hand, the pay as you go cell worked where there was no Internet connection available.

Had I been on a vacation where I had a steady Internet connection, OnSIP would have been perfect (and far less expensive than pay-as-you-go international rates, which became really ridiculous once I left the U.K.). Touring the farm country of Scotland, Wales and the Netherlands just didn't fit into that profile...at least, not in 2008. The years to come will no doubt bring a different story.

It has finally happened and I couldn't be happier. When I first got my iPhone I couldn't wait for the time when I could make extension to extension VOIP calls via my iPhone. The iPhone has wifi access. All it needed was someone to put some SIP software on the phone and, combined with Junction Networks OnSIP hosted PBX, it would be ready to go.

That time has come. Over the weekend I downloaded fring and after just a little tweaking I got it to work. The device registered to Junction Networks and my first call was to extension 7008: Tim in Chicago. All I had to do was bring up the 'dialer' in fring and dial 7008 and hit the 'SIP' button and it dialed Tim's extension. Tim only saw a call from me, ext. 7001. As far as he knew I was at my desk. In reality I was 30 feet away on my iPhone speaking to him over the iPhone's wifi connection. Next, I told Tim to hang up and call me right back. He dialed extension 7001 and fring notified me on my iPhone that I had an inbound call.

The calls sounded great. There was no echo and no delay. No question the call was better than a normal cell-phone phone call. I shared my story with fring and received the following response from fring co-founder and Chief Technology Architect Boaz Zilberman “fring prides itself on being a best-of-breed mobile Internet communications and community application, offering its users the ability to talk, chat and interact with fellow fringsters and all their online buddies using their mobile Internet connection. Partnerships with forward-thinking companies such as Junction Networks offer a new level of openness and connectivity and provide a glimpse of the future in which mobile Internet and traditional telecoms converge to provide as rich and flexible a user experience as possible.”

The best part is that extension to extension calls are FREE. Junction Networks never charges for extension to extension calls, and since fring calls use the wifi network and not the cellular network, it does not take up cellular minutes.

The fring dial pad interface does not have buttons for actions like 'Hold', 'Transfer' and 'Conference' so those features do not work in the fring interface. DTMF (touch tone) is disabled during a call. It would also be useful to be able to put SIP addresses into contacts and dial a contact's SIP address as opposed to being limited to dialing only numbers. Even with those restrictions, fring is a great start.

AT&T just announced that they are offering free wifi to all iPhone subscribers at any AT&T wifi hotspot. This includes Starbucks. Customers can locate Wi-Fi spots through AT&T's online tool or can locate a Starbucks using this tool.

With Junction Networks, since we allow our users to be registered on up to 10 different devices, this is a free addition to our service offering. There are no additional charges to add a device to the system. (Nor do we charge per user, so you could add a new user and setup the fring device as a user separate from your normal user as well.) All of our users can now setup their iPhone (or other mobile device) to act as their extension on their company PBX. No more 'forwarding calls to your cell phone'. Now, your cell phone IS your extension!

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Andy Ambramson
TMC