Despite rumors, FreeSWITCH still independent

June 15, 2009 by Garrett Smith

Late last week rumors surrounding the acquisition of FreeSWITCH by Barracuda Networks  through the open source telephony community like a California wild fire.

The only problem was they weren’t true.

What started as a speculative post by one open source proponent on Wednesday of last week, mushroomed over night, leading to a series of blog posts, forum chatter and tweets on Thursday and the early part of Friday morning.

Suspecting that something was not right, since Barracuda Networks doesn’t seem like a “fit” (and as a media sponsor of FreeSWITCH’s ClueCon developers conference, we figured we’d get a heads up), the VoIP Insider reached out to Anthony Minessale (the owner of FreeSWITCH) for comment.

In a breif email exchange, Anthony explained that FreeSWITCH had not been acquired by Barracuda Networks and that the whole rumor has its roots in the fact that a few months back Anthony took a position at Barracuda.

Good enough for us.

I guess you can chalk this whole thing up to a case of someone reading too in between the lines and others not reading close enough.

Disclosure: The VoIP Insider is a media sponsor of the ClueCon Telephony Developer’s Conference. For more information about ClueCon, please check out this post.

Improving business VoIP with edge devices

June 11, 2009 by Garrett Smith

Voice quality and reliability are the two things that every business should be concerned about.

As VoIP technology has continued to improve and evolve the traditional myth that all VoIP is of low quality and reliability has died down some. But it is still true that if you don’t take the proper steps, like proper network infrastructure and bandwidth availability, your VoIP quality and reliability will suffer.

And sometimes that isn’t enough. That’s were edge devices, or more commonly known as network monitoring devices, come into play.

Over at the newly minted Bandwidth.com blog, Luke Reynolds offered up an excellent post on the benefits of utilizing a network monitoring device. They include:

  • Voice Quality – These VoIP specific devices act as Quality of Service routers that actually shape traffic on your IP network to optimize voice quality. Most VoIP quality issues are due to bad IP traffic patterns on your network (someone watching a YouTube video is making you sound like a robot on the phone!). A VoIP edge device solves this problem.
  • Disaster Recovery – VoIP specific routers, like the EdgeMarc, support something called DNS-SRV records, which store paths to multiple gateways for automatic failover. If your data network experiences any issues or outages, a VoIP edge device can look for alternate network paths to maintain your voice service.
  • PBX Security – Your phone system (PBX) is simply an application running on a server… and it can be hacked (please use good password practices and change it frequently). While your network may have great security, VoIP edge devices add an extra layer by acting as an ALG (Application Layer Gateway) which aids in NAT traversal and will allow you to keep your PBX on a private IP. Additionally, the edge device dynamically opens and closes ports needed for voice traffic, so that nothing is left open when it’s not needed. That means your PBX and your network are less likely to be hacked.
  • Better Support Experience – [Your provider] support is great, but it helps when we have good data. Voice-specific edge devices provide real-time quality scores of Voice traffic on your network (using MOS scoring). This, along with the traffic data inside the box, allows our support team to better diagnose problems and get resolutions faster.

Now even though network monitoring devices deliver all of the benefits Luke discusses (and even more in certain deployments), these devices are often a difficult sell to customers.

Why?

It really comes down to price.

No, network monitoring devices are not that expensive, but it does represent an additional cost. One that customer’s almost never plan for.

Today some hosted VoIP and SIP trunking providers (like Bandwidth.com) mandate that a customer most have a network monitoring device, but a great many do not. A missed opportunity for the service provider and a potential nightmare in the waiting for the end customer.

In the end, what both businesses and service providers should consider when it comes to network monitoring devices (or edge devices) is whether the increased initial up-front cost is greater than the potential problems one will avoid by using one.

To me, at least, spending $600 to $3,000 (for a small to mid size deployment) is a small price to pay to prevent problems and improve VoIP service performance.

Do you agree?

Cisco Releases Official Asterisk Configuration Guide for SPA8800 Gateway

June 10, 2009 by Garrett Smith

In a move that I interpret as Cisco beginning to view the “open source” telephony market as a viable opportunity, Cisco today released a new application note on how to configure the new Cisco SPA8800 4FXS+4FXO SIP Gateway with Asterisk.  The document is entitled Configuring SPA8800 with Asterisk, and is intended to help position the Cisco SPA8800 an a cost-effective PSTN gateway for Asterisk deployments, as well as adding additional FXS ports.

Yep, you read that right….an application note, produced by Cisco, specifically for Asterisk users…..an exciting first I believe.

This application note includes configuration guidance for the Cisco SPA8800, Asterisk sip.conf, and Asterisk extensions.conf files. A troubleshooting section complete with sample traces showing registration and call flows is also included.

The Cisco SPA8800 is currently available to purchase at VoIPSupply.com.

Second Look: Siemens Gigaset A580IP

June 9, 2009 by Garrett Smith

First off, let’s describe the technology built into the Siemens Gigaset A580IP as well as its capabilities from a hardware perspective. The Siemens Gigaset A580 IP is a DECT 6.0 cordless handset which comes with a cordless handset, charging desktop docking station, DECT base unit, and power adapters. The cordless handset simply pairs to the DECT base unit much like your traditional analog cordless phones do (actually a similar technology @ 1.9 GHz) or your mobile Bluetooth headset would.

Here is where the DECT Gigaset A580IP differs from your traditional analog cordless phone; the base unit has both an analog RJ-11 port to connect your existing landline in (POTS/ PSTN), and also an RJ-45 LAN Ethernet port to plug into your IP network to support VoIP service. The base has the ability, if you configure it to, to perform auto-PSTN fall back or failover. What this means is that if you have a VOIP service or are connected to your organization’s IP PBX via SIP, and your IP connection or Internet goes down, the Gigaset will automatically route calls through your landline connection plugged into the back of the DECT base unit. Pretty effective if you are one of those people that needs to make and receive calls 24/7.

The DECT base unit has the ability to support up to 6 different cordless handsets and each can be configured as an individual SIP VOIP account. Pretty cool for a small office setup!! Please note that the product ships with only 1 handset and you can purchase up to 5 additional handsets to add on.

If you are used to IP web GUI based configurations, the Gigaset A580 IP is right up your alley as you can easily browse to its LAN DHCP fed IP address from any computer on the same LAN by simply typing its IP address in your web browser address bar.

Here are some of the first-look highlights and configurations I have performed on the A580IP:

  • Has the ability to configure up to 6 SIP VOIP accounts (I have configured 2, one for my lab Trixbox, and the other on our production Switchvox SMB. (Click Image Below for Expanded Screenshot)

  • -Easy to setup SIP registrations: (Setup for Trixbox account) (Click Image Below for Expanded Screenshot)

  • -VoIP Provider auto-configuration (auto-provisioning support)
  • -Configurable auto-PSTN fall back or failover (This is discussed above). Configurable incoming and outgoing call rules specific to the SIP VOIP connections or PSTN Landline connections. (Click Image Below for Expanded Screenshot)

  • -Configurable Dial plans for cost based routing. (Click Image Below for Expanded Screenshot)

  • -Configurable VM boxes for each SIP VOIP account (Click Image Below for Expanded Screenshot)

  • Configurable Audio codec’s per each SIP VoIP account in cases where there may be a small amount of bandwidth (G.729 codec) or to experience HD Voice (G.722) HD endpoint to HD endpoint only. Call VoIP Supply for a complete listing of HD IP Phones. (Click Image Below for Expanded Screenshot)

  • -Web Based Software Upgrades (Click Image Below for Expanded Screenshot)

Cisco Systems unveils new VoIP phones

June 8, 2009 by Garrett Smith

cisco-unveils-new-voip-phones

Earlier this month networking and enterprise VoIP giant Cisco Systems announced a brand new series of VoIP phones – the 6900 series.

First you’ve heard? Don’t feel bad.

The 6900 series seemed to have been launched with out a powerful marketing push. After diving in deeper, I found out why.

The new 6900 series of VoIP phones from Cisco are an interesting move for the company as they seem to buck a number of trends. Not just for Cisco, but the VoIP industry in general.

Upon first sight you’ll notice something different about these new VoIP phones. They’re ugly. Gone is the slick styling that businesses of all sizes have grown to crave, replaced by a rather basic and drab aesthetic.

(more…)

First Look: Siemens Gigaset S675IP / A580IP DECT VoIP Phones

June 4, 2009 by Garrett Smith

Uber-Popular in Europe, Siemens Gigaset Product Division is finally bringing several of their slick Wireless DECT hybrid PSTN-VoIP phones across the pond to the US marketplace.

The Gigaset S675 IP is comprised of a stand-alone DECT base station with integrated RJ45 ethernet port as well as an RJ11 FXO port for a PSTN landline. Plug the base station into your home of office LAN, and you can then register up to SIX individual VoIP provider accounts. The majority of users have a single account, but with multiple SIP registrations you can add additional provider accounts, or even register your S675IP as a remote extension off your Asterisk IP PBX.

If you want to maintain a landline, you can integrate it easily with the Gigaset S675IP (more…)

Don’t block Mobile VoIP, just charge more for it

June 3, 2009 by Garrett Smith

That’s exactly what Duetsche Telekom, T-Mobile parent and one of the world’s largest mobile operators, is doing.

Duetsche Telekom, which until today had banned access to VoIP services over their wireless networks, has announced new plans that will allow it’s users to pay an additional fee for the ability to place VoIP calls. The plans, which start at EUR 9.99, will be tiered based on usage.

This news is a victory for consumers (and mobile VoIP providers) who’ve been snubbed by carriers and a sign that the “tides are continuing to change” with respects to mobile carrier bans on VoIP services. Some may argue that VoIP is just an application or utility, like Google, and thus shouldn’t have to pay extra to use it.

I say take what you can get.

And while it’s still not the end goal, paying $10 for access is a small price when compared to the fees one could rack up making international calls at Duetsche Telekom’s rates.

VoIPSupply Labs: Tip for Cisco IP Phone Owners Migrating to SIP Version 12

To all Cisco CP-7940G/60G owners thinking about upgrading to Cisco’s new SIP Version 12.0 firmware….please pay attention and avoid a migraine!

Being that I run a Cisco CP-7940G on my desk with SIP protocol (under Smartnet Maintenance Contract), I received a notification in my email inbox sometime last week from Cisco telling me that the latest version of SIP protocol firmware for my CP-7940G was available for download from my CCO login on www.cisco.com.

Always wanting keep my phone up-to-date, I pointed my browser to the Cisco website and downloaded the latest version of SIP Firmware.  Once downloaded, I was quickly frustrated to find that I could not get my Cisco CP-7940G phone to take the new firmware, (more…)

First Look: Cisco SPA8800 Analog VoIP Gateway

June 1, 2009 by Garrett Smith

Cisco is set to release the Cisco SPA8800 IP telephony gateway, featuring 4 ports of FXS and FXO connectivity. The Cisco SPA8800 is an affordable IP telephony gateway solution that is ideal for small business environments that have VoIP service and also require a connection to the public switched telephone network (PSTN). It can be used with existing analog telephones and teleconferencing equipment, enabling small business customers to protect and extend their investments.

The Cisco SPA8800 features four onboard RJ11 FXS ports for connecting converting analog endpoints such as phone or fax machines to IP. In addition, the Cisco SPA8800 has four onboard RJ11 FXO ports for integrating up to 4 traditional, analog POTS lines with a SIP based IP PBX, such as Asterisk. In addition, the Cisco SPA8800 features a convenient amphenol connector interface on the back of the gateway which allows you to easily connect the unit to a telco punch-down block or mini RJ11 patch panel.

 

 

Please visit the VoIP Insider in coming weeks for more in-depth coverage of the Cisco SPA8800. The Cisco SPA8800 is currently available for purchase.

 

 

 

 

 

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