charlotte's blog

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!

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.

This morning I saw this excellent blog post by Garrett Smith which points out a few tips on making the transition to VoIP easier for SMB.

His main point is bandwidth, bandwidth, bandwidth. Although it seems obvious, adequate bandwidth is an absolute requirement for VoIP. VoIP is very sensitive to packet loss and latency, which results in quality of service issues (i.e., echo, dropped calls, etc.). Do all of your employees stream music while checking their e-mail and utilizing web applications? That's going to affect the quality of your phone calls if you don't have the bandwidth to be able to handle it.

Fortunately, in modern computing, bandwidth is one of the cheaper things that you can buy, so resolving this problem is potentially pretty easy. (Who remembers paying $1,500+/month for a 1.5 MB T1 a few years back? Good riddance to that!) But Smith's advice of doing a network analysis before buying into VoIP is super important - and easy to forget when you're excited about saving money and integrating your phone into your business life in a way that only IP telephony can do.

But it's not just the Internet bandwidth that matters - the speed of your LAN also counts. If you're using significantly older equipment, you may not have the throughput locally to handle additional traffic. If you have any equipment that is still 10baseT (and we've seen this), you probably already have speed issues with your existing data connections. Adding VoIP will be a disaster.

But there's good news here too - if you've built or upgraded your network in the last 5-8 years, it's pretty unlikely that that you're going to run into that problem. If you've signed your contract with your ISP within the last 5 years or so, you probably also have adequate bandwidth. But it certainly never hurts to get some numbers into your hand before you make the leap and add your telephony traffic to your data connection.

VoIP is excellent in many regards - but like any technology, it must be implemented correctly.

When we heard about an Open Source router at Junction Networks, we were naturally very intrigued. We love the Open Source movement and have invested heavily in it, so being able to recommend an open source router would have made us very happy.

Alas, we cannot. At least not an out-of-the-box Netgear WRG614L, which is the router touted on myopenrouter.com. We have a historic problem with Netgear routers in that they implement an ALG for SIP that cannot be turned off. (What's an ALG? Read about it here.) Unfortunately, out of the box, the WRG614L continues to have this problem, so it will not work successfully with the OnSIP Hosted PBX. We chose to test the Netgear of all the open source routers because we'd hope this had been fixed.

However, being open source, it's entirely possible for someone to change the WRG614L so that the SIP ALG can be turned off. Have you done it? We'd love to hear about it.

The Wall Street Journal has published an article today about the growth of VoIP in Africa. Research and Markets has a full study available for purchase, but the Wall Street Journal reported some highlights.

Most significantly, the growth of VoIP is stunning, with the study reporting over 100% growth in VoIP usage per year. VoIP makes a lot of sense in Africa, where affordable long distance telephony is a problem and much of the existing infrastructure is government owned and regulated. To wit, VoIP is illegal in 36 African countries, because it is driving down the prices for traditional phone calls over the existing PSTN. There are VoIP providers that operate completely illegally, which says a lot for the demand. The Global Technology Forum has an article detailing which countries had legalized VoIP as of March 2007.

Can VoIP change the world? It certainly seems so.

I'm afraid this post has nothing at all to do with VoIP. However, like the rest of the world, Junction Networks is deep in the grip of iPhone mania.

Yesterday, the Engineering department had a physical engineering problem to solve. We ordered a white board to put up in our conference room. We needed to put the white board on the wall. And, of course, we're unapologetic geeks, which means we're chronically unprepared for dealing with such mundane real world challenges.

We didn't have a tape measure. Or a hammer. But we did have several willing bodies and two iPhones with the Dual Level program, which uses the tilt functionality of the iPhone to tell you if your iPhone is level. So we stuck the iPhones in the tray of the white board and lifted it up. We made marks on the wall. We got a screwdriver and pounded some nails into the wall with the blunt end. Our white board is now being frantically scribbled on, which is a much more comfortable endeavor when it's on the wall and not sitting on the floor.

O iPhone, you are a simply brilliant device. Your possibilities seem endless.

As for my personal use, the only thing that I haven't been able to find is a Tasks program that rivals the task management on Palm OS or an Outlook/Blackberry solution. I, along with lots of other netizens who have posted about their frustration, would just love an app that synced with Apple's Mail and Calendar Tasks. Recurrence is a big missing feature in the many task programs that are out there.

But it's hard to actually want to look at things in a To Do list when there are so many other things the iPhone can do. I've been reading books with Stanza, playing games that I haven't seen since childhood, updating the social networking sites I'm involved in (which I rarely logged into before, but now update almost daily since I can do it while waiting for my train), giggling at the portable LOLcats and using aSleep as a white noise generator and meditation aid. I've spent about $10 in the App Store since I started using it, which feels like a steal for all the extra functionality I've added. And this is all in a device that's also a MP3 player and a phone.

The sheer creativity of the apps in the App Store have been really inspiring - I certainly never thought I'd be using a phone as an accurate level. I can't wait to see what else people think up to turn my phone into the best Swiss Army knife I've ever owned.

Although Cisco makes a nice line of phones, we don't recommend them for our customers because Cisco does not plan to continue to support SIP on them. They will continue developing SIP support on their Linksys branded phones, which we've tested out and really like.

However, as a courtesy for customers who already have Cisco phones, we published a Knowledge Base article on how to get them set up and configured for our OnSIP Hosted PBX, which you can find here.

Note: We were wrong. Cisco will continue to develop SIP for their Cisco line of phones. However, the Linksys phones are easier to configure and manage, so we still recommend avoiding the Cisco line if you have not already purchased your phones.

If you have an Aastra 480i, you may have noticed the note in our Knowledge Base about a bug where the 480i stops sending or receiving audio on transfers across NATs to certain phones. With the currently available firmware, the 480i writes the Refer-To in the SIP REFER packet to the user@IPAddress, which works just fine on a LAN, because the IP address is always available. Once the Internet and a NAT is introduced, this fails, since the two phones are no longer on the same network. Certain phones from other vendors can compensate for the problem, but we found that Linksys and Grandstream phones do not. The Aastra 5xi series of phones do not have this problem.

We've been working with Aastra to get this bug resolved. (It must be said that they were very responsive once we reported it - the delay in testing it was entirely ours.) They've released a beta firmware update (1.4.3), which we tested. It resolves the problem nicely, so we'll be updating our Knowledge Base as soon as it becomes generally available, which should be in the next week or so.

Lately in the Junction Networks labs, we've been focusing on IP PBXes. In that spirit, we have an Edgebox network appliance and have been testing it out.

The Edgebox is much more than a PBX - it also includes the functionality of a firewall, router, print server, file server, mail server, DNS server, etc. It can do QoS and WiFi and NAC. It's likely that I'm leaving a few other things in there - our focus was mostly on the IP PBX. Everything is administered through an attractive web interface that is intuitive and easy to use.

The IP PBX is an Asterisk based system, though you might not know that as an Edgebox administrator unless you happened to be familiar with Asterisk first. I had the Edgebox set up and connected to our PSTN gateway service within minutes. (Knowledge base article here.) I assigned it a phone number via our OnSIP interface and dialed it. Dialing out was slightly more complex - I had to create a dialing plan and give the phone I was dialing from permission to use it. That still only took a minute.

So it's a great device for our PSTN Gateway service. But there is so much other functionality in the box that OnSIP customers might want to use it for its other servers, so we tested the OnSIP Hosted PBX service through it, using it for its router and firewall functionality. There were no problems there, either. It seems to be a very solid device that is ideal for the SMB market. It definitely gets our approval.

Lately, there's been a lot of buzz in the VoIP community about high-definition telephony. The first time we heard about it, we all raised our eyebrows since high-definition is a term that gets overused in technology. But we're open-minded geeks, so we tried it...and it's really, really cool. The audio has so little distortion compared to a traditional telephony call that it's slightly unnerving at first because the vocal clarity approaches stereo quality.

So what is it? High-definition telephony comes down to the G.722 codec. In pragmatic terms, the codec is responsible for the compression and decompression of the media stream (i.e., your voice) that comprises a VoIP phone call. One of the first things a SIP phone does when it talks to a SIP proxy is to negotiate which codecs both support so that when the media stream comes, it is compressed and decompressed at the same rate and overhead. G.722 is one of the most efficient codecs out there, so it does a very good job. Voila, high-definition.

(For more information on codecs, Voip-Info has a thorough list.)

The OnSIP Hosted PBX already has high-definition functionality for extension-to-extension dialing. Both parties will need a phone that supports G.722, such as the Polycom IP 560 or the SNOM 320. (Please note, the Grandstream 2000 has support for G.722, but it's problematic.)

But we won't stop there, of course - we're currently testing moving some of our other applications to high-definition. We'll announce their launch in this blog, so stay tuned!