John Riordan

iNum == Voxbone

All OnSIP users can dial any SIP address for free. It has always been that way.

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OnSIP in The Cloud? Technical Detail About OnSIP Infrastructure

From "Cloud Computing" on Wikipedia...

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Nectar Announcement at VoiceCon and Session Border Controllers

Last week, Nectar announced the release of their Virtual Session Border Controller service (vSBC).” Nectar is notable for one of their clients’ phone service success, 1-800-FLOWERS. Based on their "Phone Options", it appears Nectar offers a hosted Avaya IP PBX service on a per seat basis. (Avaya is a Lucent spinoff and is a well known company who sells all sorts of telephone related technology, including IP PBXs.) When we came upon this announcement in VoiceCon news, we decided to write a blog on Session Border Controllers. As some background, Session Border Controller (SBC) is something you might need to connect SIP Trunks to your IP PBX, depending on your IP PBX. Many IP PBXs have the functionality built in (Asterisk, for example). Over the last ten years, the acronym SBC has been popularized by companies selling products to telephone carriers, but the functionality is simply that of a SIP back-to-back user agent (B2BUA) configured as a VoIP gateway. SBC gets in the middle of calls between two networks. For large enterprise SBC users, the logical trend seems to be to move the functionality to the routers, doing away with the SBC altogether. A recent white paper by Juniper covers the issue in some detail. In addition, Cisco offers integrated SBCs. Basically, SBCs are typically located at the same demarcation point as routers/firewalls, so integration seems like a sensible approach to reduce the number of parts to buy and operate. Our position on SBCs is that they are not true peer-to-peer networking. They get in the way of end points talking to each other directly, adding a "hop" for the media to go through and potentially degrading performance. In some instances, they can prevent end-to-end encryption from being used. If you are implementing a new VoIP set-up, we don’t recommend using SBC. However, if you are using a PBX box requiring SBC, hosted SBC may be an option for you.

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RFC 5589: Good news for SIP Transfer

While the REFER request has been around since RFC 3515 (April 2003), the details on how to actually implement call transfer have been somewhat grey. In practice, this has lead to less than stellar interoperations between vendors' phones. For example, Polycom phones have a "semi-attended transfer" implementation, enabled by default, that isn’t supported by phones from other manufacturers. While my opinion is that this should be disabled by default, I have sympathy for Polycom's dilemma: they wish to offer this functionality to their users (common on traditional PBXes), but there has been no standard way to do so. As a side note, our boot server disables Polycom's semi-attended transfer feature for OnSIP users.

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Why we don't run our systems "In the Cloud"

I occasionally get questioned about our choice to run our own network and servers when we could outsource it all to a "cloud" provider like Amazon and thus save a lot of money while simultaneously improving the reliability of our service. I hate this question because it is loaded, and I know they are not likely to understand why we can't run on a cloud service no matter how patiently I try to explain the technical issues.

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Cordless SIP Phone - Aastra 57i CT

I don't currently have an official home office, but I've had an office phone at home for years now. This home office phone mirrors the phone on my desk at work. Both phones are registered as a contact for my SIP address so they both receive all my calls an otherwise behave exactly the same - all very easy to do with OnSIP Hosted PBX, but that's another story. A few weeks ago I swapped out my home office phone for an Aastra 57i CT which comes with this nice little cordless handset that syncs with the main station.

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Topics: phone reviews

Detecting IP Packet Loss for UberNerds

I spent the good part of a couple of days last week helping track down packet loss at the edge of a PSTN carrier's IP network. They were dropping 1-5 packets out of every 10000 due to an incorrectly configured port on a gigabit ethernet switch.

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Skype Not Worth A Gazillion Times Earnings? Say It Isn’t So!

On and off over the last couple of years, as Skype, YouTube, and more recently, Facebook, got tagged with silly sums, and I've been having the occasional flashback to bubble-land circa 1999. The hype, the parties, and that good old frothy feeling. Ah, the memories.

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Iperf - Simulating VOIP Traffic for UberNerds

Have you ever wanted to hammer a part of your network or a particular system to get an empirical view of its performance characteristics under a VOIP load? Are you someone who is compelled to understand how things work in grim detail, even if the things in question are tedious in the extreme - like network performance measurement? If not, don't feel bad. As it was explained to me, "Not everyone who visits the blog is an UberNerd, or aspires to UberNerdity, but on the other hand those who display UberNerditude in the comment threads are treated with a respect bordering on lunacy. That’s just the way we are."

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