Tech Talk- IT & Code

03
OCT 2008

Posted by Charlotte Oliver at 02:03 PM EDT

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Junction Networks Labs: My Open Router

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.

24
SEP 2008

Posted by Charlotte Oliver at 02:52 PM EDT

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iPhone Mania Hits Junction Networks

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.

19
SEP 2008

Posted by Charlotte Oliver at 10:42 AM EDT

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Junction Networks Lab: Cisco 7960 Phones

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.

17
SEP 2008

Posted by Charlotte Oliver at 10:58 AM EDT

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Aastra 480i Bugfix - More from the Junction Networks Lab

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.

29
AUG 2008

Posted by Charlotte Oliver at 02:22 PM EDT

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Junction Networks: EdgeBox

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.

27
AUG 2008

Posted by John Riordan at 11:28 AM EDT

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Voodoo - Freeze Your Hard Drive to Recover Data

While a true story, the following contains what may very well be a bad idea... I have an external 250 GB hard drive that, as of a few days ago, would not spin up. Upon power up, it would make an short buzzing sound following by, click, click, click, click, click and so on. I tried disconnecting and reconnected it a few dozen times with different cables, different ports, different power supply. No luck. Finally, I decided to try freezing the drive by letting it sit on top of a large gel ice pack for 15 minutes. While still sitting on the ice pack, I powered it and it spun up on the first try! I then immediately began copying everything off the drive. The drive ran for about 3 hours while the ice pack slowly melted. Then it was back to click, click, click... I got the idea from here: http://www.macosxhints.com/article.php?story=2006110111270170

21
AUG 2008

Posted by Charlotte Oliver at 02:51 PM EDT

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On Codecs and High-Definition VoIP

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.

06
AUG 2008

Posted by Charlotte Oliver at 04:37 PM EDT

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Response Point and Junction Networks

Some of you may have seen the press release from Microsoft announcing the release of the Response Point PBX and the role of Junction Networks as a preferred service provider. Our favorite write-up of the partnership was over at Asterisk VOIP News, though that may be because they included a glowing note from the editor about the reliability of our service. We tested the Response Point in our labs, of course, and found the setup to be almost ridiculously easy. The test network was set up and ready to go within 30 minutes of unpacking the box the system came in. Most of the Response Point customers that we've spoken to report a similar experience.

29
JUL 2008

Posted by Erick J Johnson at 05:28 PM EDT

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A VoIP Web Service API for Everybody

Did you know that Junction Networks has a public web service API available to any of our customers that exposes all of the pieces necessary to manage your own hosted PBX and PSTN gateway services? In fact our admin.onsip.com web administration portal has been built entirely on top of the very same API that is open and available to the public - this means any feature you see in our administration portal is potentially available for you to implement in your very own VoIP product.

18
JUL 2008

Posted by Charlotte Oliver at 08:50 PM EDT

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This Week in the Junction Networks Lab - Application Level Gateways and More!

We've had a really busy week in the Junction Networks lab. We've started a testing program for some of the routers out on the market so that we can come up with some solid recommendations for routers that are known to work well with our network. (Alas, not all routers are created equally.) The main rule is that the router needs to not interfere with SIP. Specifically, we need routers that don't rewrite SIP packets, because we do a fair amount of work in learning about the network that a packet comes from in order to route and transfer calls properly.

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