Will high definition or standards revive voice communications?

April 13, 2009 by Garrett Smith

Daniel Berninger penned a guest piece on Jeff Pulver’s blog this morning entitled, “The HD Connect Manifesto” which anoints high definition voice as the savior of the voice communications industry.

In reality the manifesto has little to do with “HD” and more with two enablers driven by standards:

  • Convenience
  • Interconnection

Make voice communications more convenient

Today communications is increasingly fragmented. Consumers and businesses have more choices.

Like water people tend to take the path of least resistance. This means that they will naturally gravitate towards products and services which are more convenient and easier to use. A prime example of this in the communications sector is the increased utilization of contextual forms of communications (email, IM, SMS).

While you could argue this point, the usage data shows that contextual forms of communications are a more convenient form of communications. Further proof can be seen in their increased utilization over time.

HD voice does not make voice communications more convenient – the fact that it is a standards based technology mean that it has the potential to make voice calling more convenient.

Take the dreaded phone number. Daniel points out that the adoption of SIP URI’s by a voice service provider (instead of traditional phone numbers) would provide unified framework and therefore add conveniences to a voice service.

This certainly is true, but it’s not as a result of HD voice. It’s thanks to SIP, a standards based protocol.

Make voice communications more efficient (through interconnection)

Today most VoIP service providers are held captive by those who own/control the last mile and of course the PSTN. It adds inefficiencies to their services. This allows other voice service providers to continue to compete with them. It is, as Daniel states, the “Faith in the status quo…” that keeps them ticking.

That is why the idea of an interconnected network of ITSP makes sense. Creating a separate, more efficient network from which to power high quality voice calls is a great idea.

The yet-to-be-created network would increase efficiencies leading to increased flexibility and further cost reduction for consumers. But HD voice is not driving this.

The utilization of standards based technology that makes this possible.

HD voice is great, but standards are the true drivers

HD voice is a great thing for the voice communications industry. It is does provide call quality that is superior to the PSTN, however better sounding phone calls is not going to lead a resurgence in the voice industry.

What will lead the resurgence of the voice industry is the use of open, standards based technology to make communication more convenient, efficient and cost effective – whether those standards include high definition codecs or not.

And the winner is…

April 10, 2009 by Garrett Smith

Drum roll please.

The winner of the “what’s important in a business VoIP service” contest is commenter #10 Fred Posner.

Fred’s number was drawn out of a hat just a few minutes ago and he will be the recipient of the QuickPhones QA-342 wireless VoIP phone.

Thanks again to everyone who participated in the contest and Fred I’ll be dropping you an email later tonight or tomorrow to get your shipping details.

Last chance to win a FREE wireless VoIP phone

Last week Friday we launched a contest for readers giving the a chance to win a wireless VoIP phone for answering the question:

“What’s important in a business VoIP service? (Full post here)

Well today is the last day to get in on the action as we will be picking one comment at random to win the QuickPhones QA-342 at 5pm today.

So if you’ve got thoughts on what makes a business VoIP service great take a minute to leave a comment and earn your chance to win that FREE wireless VoIP phone.

Polycom KIRK 2010

April 9, 2009 by Garrett Smith

Polycom has announced the latest addition to their DECT product family, the KIRK 2010.

The Polycom KIRK 2010 is a DECT based wireless VoIP phone. At the surface it looks just like another slick wireless VoIP phone, but the most important aspect of the KIRK 2010 might the targeted customer:

The SMB.

If you’re familiar with the Polycom KIRK line of wireless VoIP phones you know that they’re excellent – with the exception that they are typically “outside the reach” of small medium businesses.

In the KIRK 2010, it seems that Polycom has constructed a simple, basic wireless VoIP phone that will allow the small medium business leverage the benefits that come with mobility.  This simple and basic theme can be seen in the KIRK 2010’s feature set:

  • Black & white LCD screen (3 lines of text/icons)
  • Internal/external ring pattern, volume control and silent modes
  • Telephone book with room for 40 numbers
  • Speech/stand by time > 12/150 hours
  • Weight incl. battery: 120g
  • Size (LxWxH): 124x47x31mm

There’s no word yet on what the price tag of this unit will be or when it will be available, but we will keep you updated.

Adding Telephony to iPhone Applications

One of the great things about the iPhone is the ability to add a variety of different applications. But outside of the applications for mobile VoIP providers, there aren’t too many iPhone applications that feature telephony capabilities.

Now this is likely due to the fact that iPhone developers have no clue when it comes to telephony. That’s why I thought it would be great to share a session Ifbyphone CEO Irv Shapiro put on at the recent ecomm conference.

Entitled, “How to build an Ifbyphone-to-iPhone app” the session shows iPhone developers how to use Ifbyphone’s cloud telephony capabilites to add voice to their applications. What’s even better is that the examples and advice come from Ifbyphone’s own experience.
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Digium Releases Switchvox 4.0 (Finally!)

April 8, 2009 by Garrett Smith

Switchvox SMB release 4.0 shipped yesterday….we upgraded our SMB plant here last night without any real hiccups. With the release of SMB 4.0, a great IP communications platform just got better in many ways.

1 – Switchvox SMB now supports “Presence” across multiple sites (server instances). This was an issue with our own deployment in the past, and it is great now that you can unify multiple, geographically dispersed Switchvox SMB systems with a “global” Switchboard, showing you the status of all users and allowing you to control calls across multiple sites.

2 – Fax! Switchvox SMB 4.0 now supports outbound fax and fax-to-email. I have not had a chance to test this personally but am looking forward to accidentally knocking our finicky analog fax machine off the desk soon.

3 – More API goodness! Digium/Switchvox are wisely investing dev cycles in their API Call Control, unlocking more and more flexibility and integration abilities.

4 – Jabber/Chat capabilities with Switchboard. Perhaps soon I can ditch Windows Live Messenger and AIM in favor of unified IM/Chat within Switchboard.

5 – More / Better Microsoft Outlook integration!

6 – Improved system diagnostics and schedule reports.

There’s more to love in Switchvox SMB release 4.0 beyond the features I’ve mentioned here. If you are considering Switchvox or have any questions about this product, feel free to contact one of our product specialists at 800.398.8647!

Polycom Releases New SIP 3.1.2 Firmware for SoundPoint IP Phones

April 7, 2009 by Garrett Smith

Polycom has just released new SIP firmware version 3.1.2, and it can be found here. This release is identical to the SIP 3.1.2 release with the addition of software and configuration parameters applicable to the VVX 1500 phone.

Here are a few of the highlights:

  • SIP 3.1.2 includes GA level support for SoundStation IP 7000 integration with Polycom HDX video systems. This feature requires that the SoundStation IP 7000 is running BootROM 4.1.2 or newer software.
  • The SoundPoint IP/SoundStation IP XML API feature is now formally supported in this release. This feature was designated as a Beta in the SIP 3.1.0 and SIP 3.1.1 releases.
  • SoundPoint IP 550, 560, 650 and 670 products require BootROM 4.1.0 or newer in order to load SIP firmware release 3.1.2

If you are considering upgrading, make sure you have the correct bootrom, depending upon the Polycom SoundPoint model you own, as follows:

Digium Adds PSTN Faxing Support to Open Source Asterisk

April 6, 2009 by Garrett Smith

Fresh off their announcement of revamped, comprehensive support package offerings for open source Asterisk, Digium today has released Fax for Asterisk .

HUNTSVILLE, Ala.—April 6, 2009—Digium®, Inc., the Asterisk® Company, today announced Fax For Asterisk, a complete, cost-effective platform for the development of fax solutions. The offering provides Asterisk users and integrators a suite of user-friendly applications and a licensed version of the industry-leading fax modem software from Commetrex. To meet the demanding requirements of business users, Fax For Asterisk provides reliable faxing across the Internet and public switched telephone network (PSTN).

Asterisk is the most widely used open source telephony platform. The software is available free of charge and has been downloaded millions of times for use by individual developers and systems integrators creating custom telephony solutions for businesses. Asterisk is also available as the professional-grade and commercially supported Asterisk Business Edition.

“Asterisk users, developers and integrators now have a toolkit allowing them to integrate fax with their phone systems,” said Bill Miller, vice president of product management at Digium. “With Fax For Asterisk, Digium offers a reliable and fully supported fax solution.”

Fax For Asterisk interoperates with standards-compliant fax machines connected to Asterisk 1.4 and 1.6 on x86 Linux systems. It provides low-speed PSTN faxing via DAHDI-compatible telephony interface cards as well as VoIP faxing to T.38-compatible SIP end points and service providers. Fax For Asterisk operates at speeds up to 14.4kbps and supports V.17, V.27 and V.29 fax modems.

Fax For Asterisk is available free of charge from the Digium webstore at http://store.digium.com/ for one concurrent fax session. Multi-session licenses are available for a one-time fee of $38.50 per channel. Fax For Asterisk is available immediately. Fax capabilities for Digium’s Switchvox IP PBX were announced in February of this year and are based on this solution. For more details, visit www.digium.com.

Fax solutions for Asterisk are not new (Hylafax, SpanDSP, etc) but direct support from Digium for faxing is, and this plugs another hole in open source Asterisk making it an even more compelling option that has the Shoretel’s of the world looking in their rearview mirror.

The six month VoIP system ROI

This morning Doug Mohney at FierceVoIP dropped some great observations from last weeks VoiceCon show. The one observation that struck a cord here at VoIP Supply was the six month ROI (Return on Investment) businesses want on the purchase of a VoIP system.

A year ago businesses were happy with an 18 month ROI. With new financial pressures (thanks to the economy) an 18 month ROI is no longer realistic for them.

But is a six month ROI realistic either?

Well, let’s take a look.
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