7 Ways To Improve VoIP Call Quality

February 23, 2007 by Garrett Smith

VoIP Call Quality is the Most Important Factor For Today’s VoIP Consumer

Tracy Mayor over at VoIP-News have an excellent post on seven ways to improve VoIP call quality. Here are Tracy’s top 7 ways to improve VoIP call quality:

1. Go private rather than public. “The public Internet is always going to be more subject to disruption and therefore to quality issues than private networks,” points out Jan Dawson, vice president of US Enterprise Practice at Ovum, a Boston-based telecom consultancy. “The best VoIP services use private, dedicated networks rather than the public Internet.”

2. Better cables equal better performance. For many companies, only CAT-5 cabling on-premises is robust enough and fast enough to handle VoIP traffic adequately.

3. Hash things out with your vendor ahead of time. “The most important question to ask is how will you care for your VoIP infrastructure on Day 2,” the day after installation, says Pierce. “Most customers don’t think about per-call monitoring, call management or trouble-resolution until later, and they get themselves into deep trouble that way.” To avoid the same mistake talk to your vendor . Ask your vendor what, if any, call- and network-monitoring tools they include or at least support. With service providers work up service-level agreements (SLAs) that specify what quality-control metrics will be used to measure jitter and packet loss and how such problems will be addressed.

4. Allocate bandwidth wisely. Separate is better, says Dawson, meaning companies should consider using either two separate LANs for voice and data or keeping traffic virtually separate through use of a VLAN. Full duplex, non-blocking switches will help to avoid collision and packet loss. Further, make use of network analysis tools to identify and eliminate congestion points on the network.

5. Buy VoIP hardware, handsets and the IP PBX, with built-in echo cancellation. The longer the echo cancellation “tail length” (measured in milliseconds), the more effectively the technology will work.

6. Upgrade your existing network management and monitoring suite. They should accommodate VoIP, which includes upgrading your RMON (remote network monitoring) probe and protocol analyzers to recognize and decode VoIP traffic.

7. …Or bring a network monitoring system online for the first time. The ultimate goal should be a system that’s capable of proactively monitoring each call from SIP phone to SIP phone across the WAN, says Pierce. While she warns “there’s no one single off-the-shelf package” that can answer all your VOIP monitoring needs, good places to start are with network monitoring suites from your IP PBX vendor or VoIP-accommodating network-monitoring suites like those from HP, Fluke/Visual Networks, Psytechnics and IBM’s Tivoli division, which now owns the Netcool product line formerly sold by Micromuse.

I would like to add few other ways that you can improve your VoIP call quality:

  1. Don’t Oversubscribe. Most businesses try to make too many calls over to small of a pipe. Bad idea. Make sure you have adequete bandwidth available for the amount of calls you need to make.
  2. Upgrade your network infrastructure. Surprisingly, the cost of upgrading your network infrastructure is cheaper than ever. For example, 24 port PoE switches can now be had for around $399. If your network is not in tip-top shape, you run the risk of having packet loss.
  3. Experiment with Codec Size.If you have enough bandwidth, try using a large codec. Although your service provider might compress this on their network, it should not affect the decompression of the codec on the LAN. Alternatively, you could pick a service provider that uses a larger codec by default.

GrandCentral Now Interoperable With Gizmo Project

February 22, 2007 by Garrett Smith

Now This is a next generation voice partnership

GrandCentral, the company that allows your single phone number to have unique features and control capabilities, and Gizmo Project, the popular SIP business VoIP service have announced that their services are fully interoperable.

So What’s The Scope?

GrandCentral customers can now designate their free Gizmo Project profile ID as a destination numbers which will ring on their personal computers or select next generation Nokia dual mode Nseries mobile phone or Internet Tablet, whenever a call comes into their GrandCentral number.

Prior to this announcement, their was no way to link your Gizmo Project profile id to your GrandCentral account. This new capability will also allow Gizmo Project users to recieve free calls directly from the PSTN. Not only does this interoperability further the functionality of Gizmo Project, it also eliminates the need for Gizmo Project users to purchase a seperate DID for their Gizmo project account, a savings of around $35. Not too shabby!

Exclusive Interview With Vocalocity’s David Politis

February 20, 2007 by Garrett Smith

Hosted IP PBX Solutions Are Growing In Popularity With Small Medium Businesses

In-stat, a market research firm, projects that by 2010, there will be more then 3 Million hosted VoIP seats in the US alone, up from 373,000 in 2006. Fueled by cost savings, the small business market continues to be the hot zone for hosted providers with most deployments in the 20-to-50 seat range. “As business managers assess their resident ability to deploy and manage enterprise-like VoIP services, many are finding themselves lacking the capital and expertise required,” says David Lemelin, In-Stat analyst. “As a result, hosted VoIP solutions are becoming more attractive.”

One such hosted VoIP system provider blazing a trial is Vocalocity. I was fortunate to have the opportunity to sit down last week and chat with Vocalocity’s Executive Vice President and General Manager, David Politis, about Vocalocity, their offerings, and the future of the hosted VoIP PBX industry.

Garrett: David, Give Us Some More Information about Vocalocity.

David: The story of Vocalocity starts with a company called ZivVa. In 2003 Boris Jerkunica (currently CEO of Vocalocity) and Phil Hill (currently President of Vocalocity) founded ZivVa, in Atlanta, GA, with the vision of making it easier for people around the world to communicate. The first service that ZivVa offered was an international virtual number service, that allowed immigrants or business people around the world, to buy a local number in a country and city, that they where they were not located, and forward it anywhere else in the world.

ZivVa experienced a lot of success in the Eastern European and South American markets. Then in June of 2005, developed a hosted PBX product targeted at the Small and Medium sized business called ZivVa Office. In June 2006, ZivVa acquired Vocalocity, also an Atlanta company, for its industry leading VoIP platform and have since been combining the two technologies to create a state of the art hosted pbx offering.

Since then the hosted PBX product has been renamed Vocalocity PBX. Over the last year Vocalocity has experienced tremendous growth and has become a leader in the market for small business VoIP. Vocalocity’s vision is to be the leader in innovation for small business VoIP. We continually strive to deliver the best in value, technology and support to our customers.

Garrett: Tell us a little bit more about your service.

David: VocalocityPBX is the most cost effective, reliable and feature rich hosted pbx on the market. We offer small businesses the type of office phone that used to only be available to enterprises at a small fraction of the price. Our product allows small businesses the flexibility that they need. We offer a variety of plans and our product is built so that we can grow with our customers. There is no maximum to how many phones you can have on one of our PBXs. The vocalcityPBX product is built so that is plug and play, we’ve eliminated the need for an installation. For years the phone system has been a headache for small business and we want it to be as easy as email. We can do all of this because of the power of VoIP technology and especially our proprietary platform. Businesses no longer have to have a PBX on site, we host all of that for the business and let them focus on what they’re good at, their business.

Garrett: What makes Vocalocity different from other hosted IP PBX providers?

David:
We differentiate ourselves in three areas.

  • Unchallenged Value – We are the best value on the market. We deliver the best product at the best price. The most popular plans with vocalocity are the flat rate plans that include unlimited usage, all taxes and fees are bundled into that price. You dont see that with any of our competitors, they may quote a customer $39.99 per extensions, but when the bill comes it is $49.99 per extension. Our unlimited extension is $34.99 and that is what you pay.
  • Proprietary Technology – Most hosted PBX vendors have purchased their technology from a third party and are therefore don’t have the flexibility to modify or improve on their product. They have to wait for the newest update to come out in order to update their customers PBXs. Vocalocity is constantly innovating by delivering top notch reliability and a constant flow of new features.
  • Support First Philosophy – Vocalocity is the first in the industry to introduce a Personal Account Manager for all of its accounts. Now, small businesses have an employee at Vocalocity who is dedicated to helping them with any questions or concerns that they may have. It’s like the Cheers song, its good to have a place where people know your name. The philosophy of everyone in the company from CEO to Office Manager is that our customers happiness comes first.

Garrett: Can you give all a good idea of the different plan? How Much Does a basic hosted plan cost?

We have a number of plans and add ons to fit any small business needs. Our most popular VoIP service plan is our unlimited extension plan, this gives the customer unlimited calling to the US and Canada and all of our features at a price of $34.99 per month. We also offer metered plans for businesses that dont use their phones as much, but would still want the benefit of a full featured PBX. These plans are $14.99 a month plus a per minute fee.

Garrett: Any other important value adds about the service?

David: Some of the add ons that Vocalocity offers are, Fax, Conference Bridge (up to 30 participants), Call Queuing, Virtual Departments, and many more. We have any feature that a small business would be looking for.

Garrett: So what are some of the business benefits of going with a hosted solution, as opposed to a premise based PBX solution?

David: There are a few.

  • Cost – With a hosted solution Customers can eliminate the upfront equipment costs of an expensive on premise phone systems.
  • Reliability and Disaster Recovery – Hosted solutions store all of your PBX information and files at hosting facilities that have sophisticated and redunant equipment, so that the data is secure and can always be recovered in case a customer needs to move offices or work from home. With an on premise PBX all data (including VM, settings, faxes, etc) are stored on the PBX that is on the customer’s premise. If for any reason (and there are many possibilities) they lose connectivity to their PBX they are unreachable by the rest of the world and their data is irretrievable. Also, if the PBX happens to fail or freeze the customer has lost their data and needs to invest in a new PBX.
  • Virtual Office – A hosted solution makes the idea of a virtual office truly come to light. You can route calls to any employee anywhere on earth that has a high-speed internet connection. With a hosted solution there is no need for a main office. Everyone could be a remote employee, but to the outsider, no one would know (unless you told them).
  • No Maintenance or Upgrades – With hosted solutions you have no hardware or technology to maintain on premise: all day-to-day operations and maintenance of the PBX is performed by the hosting company. Companies also gain access to the latest technology automatically, with all upgrades handled by a hosted PBX company like Vocalocity. This comes as a tremendous cost savings to the customer, and eliminates the need for an on-site PBX technician.

Garrett: What does a typical Vocalocity customer look like?

David: Our average customer is a small business with between 5-30 employees. We have seen great success with business in the real estate, law and financial services industries. We service customers everywhere from California to NY, the beauty of a hosted PBX is that it has no boundaries.

Garrett: What do you do for companies that need a fax line?

David: We offer a fax solution that customers seem to like more then the traditional fax solution. It is a fax to email, email to fax solution. Simply put, a customer can attach a file to an email and address it to the desired fax [email protected] and send the email. The fax will come out on a fax machine at the desired location. When customers receive faxes they simply come in as a file attachment to an email. They double click on it and can view it or manipulate it on their computer.

Garrett: What Do You See as the Future of Hosted IP PBX solutions?

David: The future of hosted IP PBX communications is very bright. I’m sure you are well aware that a small percentage of small and medium businesses are using IP communications today, let alone hosted IP communications. I see the space that we’re in as very similar to the CRM space, in the not so distant past companies had no choice but to put their CRM solution on premise and deal with the headaches and hassle of installing, upgrading and maintaining another piece of hardware on their network that stored arguably their most important data. Now, hosted/on demand CRM applications like Salesforce.com are becoming the preferred CRM tool for business owners. All of the data is backed up in more secure locations then a small office, the system is accessible over the web from anywhere in the world and most importantly it has become easier to integrate other business critical applications with a companies CRM. I see the same happening with business telephony.

When it comes to business VoIP, business owners and IT directors have the same expectations. It has been expected that when it comes time to buy a phone system, there are going to be messy installs, headaches with configuration and other misqueues and confusion involved. Hosted IP PBXs can eliminate all of that hassle. Most businesses already have a high speed broadband connection installed in their office and have a network set up. With a truly hosted solution like ours, the phone system just becomes another application on that network, no worries, just plug and play. This is already something that is happening today, especially in the small business. But, it won’t be long until enterprises realize the benefit and trust the reliability of a hosted solution.

I believe that the most exciting prospect for hosted business VoIP systems is the new found ability and focus on creating new unique and useful applications that will help increase worker productivity and enable integration with other business tools. All of this is going to be done while showing companies significant savings that they can then reinvest into their core business.

Switchvox IP PBX Solutions

February 19, 2007 by Arthur Miller

I have received feedback from a lot of readers regarding the Linksys Voice System blog last week. Several people found it intriguing enough to call me and have a discussion about their experiences with the system. I appreciate that. Please continue to contact me.

Switchvox SoHo or Switchvox SMB?

This week I would like to take a look at another prominent VoIP PBX, Switchvox. Switchvox has two different versions to offer consumers, the SoHo edition and the SMB edition.

There are a few notable differences between the two offerings. The SMB edition has conference bridge functionality, extension groups, the Switchvox switchboard, enhanced call que statistics and more complex IVR actions.

The SOHO has a basic conference room, will only allow you to distinguish between groups to make multiple directories, and has some call center features, but the SMB has more robust reporting and supports a less advanced auto-attendant style IVR. Should your business outgrow SoHo phone systems edition Switchvox allows you to upgrade to the SMB edition for $1,500.00.

In summary, the SoHo is meant for a really small office, perhaps looking for an auto-attendant but not looking for voice and data integration, also not needing any intercom because the other users in the office are probably going to be in ear-shot anyway.

Buying Decisions: Linksys OR Switchvox?

There are numerous differences between the Linksys LVS-9000 and the Switchvox platforms. I will mention just a few determinative features comparing the two systems. The Linksys Voice System has a sixteen user limit, the Switchvox system is uncapped. And by uncapped I mean that it is technically capped depending on your bandwidth limitations and need for QOS (which is almost always the absolute highest priority in my experience).

Switchvox systems generally work with twenty-three concurrent calls, a number that assumes at any given time some users will be taking an inbound call, others will be making an outbound call, and some will be checking voicemail. Twenty four concurrent might not sound like a lot when comparing the number of users (sixteen) from the Linksys system, but keep in mind that the numbers are different. The Switchvox system is talking about concurrent calls; the Linksys system is talking about total users. Twenty four concurrent calls is generally a number equated with business having over one hundred users. Each system also uses IP Phones differently.

Both the Linksys IP PBX and the Switchvox VoIP phone system will accept any SIP compatible phone: Grandstream, Linksys, Cisco, Polycom, Aastra to name a few. The LVS-9000 features are limited if purchasing any non-linksys phones (notably the message waiting indicator light will not work). Switchvox features are not limited by the phones; however Switchvox will not support anything other than their own pre-provisioned Polycom phones. Switchvox also offers a robust and user friendly GUI, whereas the Linksys system does not.

Benefits of Switchvox / Switchvox sold to Best Kept A Secret Technologies

The SoHo has its place, but for the relatively small price difference I find that most clients prefer to purchase the SMB edition. I had one such client conversation this week. The prospect calling in, needed an in house 15 user VoIP phone system solution, and though he fit into the user number that the Linksys system could provide my prospect was looking for a system that would scale to a larger 20 user operation. He also had a need for the easy to use drag and drop functions with a customizable interface. After discussing the benefits of Switchvox and revealing a free demo of the “switchboard” feature the customer was ready to order.

I recommended:

A one year support contract from Switchvox at $499; although the Switchvox system ships at nearly a plug and play level, the customer did not have any technical staff in house to assist with the installation and support of the system.

The customer did not need any analog phone line capability or an analog PCI card would have been suggested.

The customer was using an existing T1/PRI line so for $664 a provisioned Digium TE110P was added for connectivity to the PBX.

We discussed the differences between the IP301, IP501, and IP601. His desired price point was in the IP301 range however the customer needed the full-duplex speaker phone for each user and the Polycom IP-501 provided the best value with the features he needed at $264 ea. They also purchased a new conference phone.

Linksys CIT400 iPhone Review

The Linksys CIT400 is a Dual-Mode Dect Phone For Use With Skype

Stephen Pinches, over at Skype Gear, has an excellent review of the Linksys CIT400. The Linksys CIT400 is a Dual-Mode DECT phone that allows users to send and receive both Skype and PSTN calls, making it an excellent solution for residential users looking for Skype hardware.

According to Stephen,

“The CIT400 is perfect for people who still rely on their old-style phone network, but also want to make cheap Skype-to-Skype and Skype-to-other phone numbers calls. Additionally, the CIT400 offers access to Skype Voicemail, as well as being able to search for users, accept or decline incoming buddy requests, view the basic details of users’ profiles and also quickly and easily see how much credit you have left.”

The only major drawbacks?

“Unfortunately, Linksys have as yet not confirmed availability of additional handsets.”

The Linksys CIT400 is now on sale for $164.99 and includes the handset, charger, and a Base Station that plugs into your broadband network and to the standard telephone line plug. For more of the Linksys CIT400 review click here.

Sangoma Technology Mail in Rebates

Save up to $250 on Select Sangoma PCI Cards For Use With Asterisk and trixbox!

Sangoma is the leading provider of connectivity hardware and software products for Wide Area Network (WAN) and voice infrastructure. For a limited time we are offering Mail-In Rebates on select Sangoma PCI Cards, all Asterisk Interoperable.

Here Are the Details:

New Aastra IP Phones Have Arrived

February 15, 2007 by Garrett Smith

The latest line of Aastra IP Phones are finally available from VoIP Supply!

The newest Aastra IP Phones are loaded with new features and a completely new design. For more about the specific models, please click on the links below.

Look for more details about the exciting new phones as soon as they are made available.

To VoIP or Not to VoIP, That is the Question…

February 14, 2007 by Garrett Smith

As a salesperson, it’s ironic that I spend a lot of time convincing people not to use Voice over IP (VoIP). It’s not that I don’t truly believe in the value of IP. Rather, I am a huge proponent for the “appropriate” use of IP, where it makes business sense and offers an appreciable ROI for the user.

IP was engineered as a best-effort, data networking system with inherent disaster recovery capabilities that the PSTN lacks. Most user are, or should be, more concerned with total availability, reliability and quality of service.

The traditional PSTN (Public Switched Telephone Network) is comprised of a two-level, circuit switched hierarchy of Class 5 (Local) and Class 4 (Long Distance) switches.
The local loop, or “subscriber line”, is the physical circuit connecting the customer (user) to their telecommunications service provider network. In a traditional PSTN carrier network, this local loop is terminated in a circuit switch maintained by an ILEC (Incumbent Local Exchange Carrier). The PSTN network for transporting phone calls has remained virtually unchanged, save for the introduction of electronic telephone exchanges in the mid 20th century.

To contrast, in a VoIP network the local loop is the same as the PSTN, but the transport mechanism underlying it is vastly different. IP communications transport backbones use an IP/MPLS infrastructure, technologies that were originally conceived for the transport of data, and have been adapted for voice transport.

If we state that the primary benefit of VoIP is cheaper telephone calls, which seems to be a popular statement these days, we are failing to illustrate the larger picture. The true promise of VoIP is that it should allow for the deployment of new services not possible on the traditional PSTN, and therein lies the true potential long term benefit for the majority of consumers and business users.

On the LAN side, the internal IP network that resides within most homes and offices today, VoIP offers significant business benefit and immediate ROI for most users. If you take a look at the basic feature set of practically any VoIP PBX….flexible auto attendant, voicemail/email integration, user call control, “presence”, support for remote workers, ease of administration….these are real benefits that many businesses lack today that can have a dramatic positive impact.

On the WAN side, calls coming in going out, there is often little, if any justification for the use of VoIP. As carriers and service providers mature, and begin to offer innovative features and services not previously viable on the traditional PSTN, they will create the proper justification and true ROI for both consumers and business users, and fully deliver on the promise of VoIP.

Rhino R4FXO-EC New Analog PCI Card From Rhino!

Rhino Equipment Has Released Their New 4 Port FXO Card With Echo Cancellation
rhino r4fxo-ec

The Rhino R4FXO-EC is a next generation analog PCI card for use with Asterisk and other open source telephony platforms. The R4FXO-EC features an on-board control element and on-board Echo Cancellation Circuit. The on-board control element prevents PCI bus “bit banging” which allows for lower CPU power, while the on-board Echo Cancellation circuit provides echo protection to ensure crisp clean calls.

Rhino R4FXO-EC Features & Functionality

  • Asterisk soft PBX tested and ready
  • Zaptel-compliant open source Linux module source code
  • Proven Silicon Labs FXO DAA component – Si3050
  • Silicon Labs international line interface device – Si3019
  • Custom Rhino PCI interface chip means no excess CPU overhead
  • Four RJ11 jacks at card bracket
  • Field software upgradable
  • All major signaling modes supported
  • Loop start signaling for advanced features such as Caller ID and Distinctive Ring

Rhino R4FXO-EC Summary

The Rhino R4FXO-EC is an excellent alternative to analog PCI cards manufactured by Digium and Sangoma. With a price of $359.99, the R4FXO-EC is about $20USD cheaper than competiting cards, so for those of you on a tight budget it is certainly a available solution.

More from: Asterisk Garrett Smith

Asterisk 1.2.15 Released

February 13, 2007 by Garrett Smith

Asterisk 1.2.15 Has Been Released

The Asterisk development team is pleased to announce the release of open source PBX Asterisk 1.2.15.

This release contains a large number of bug fixes, and some significant improvements:

* Support for Zaptel-based transcoder hardware, initially the Digium TC400B 92/96 channel transcoder.
* Handling of voicemail subdirectories when using ODBC storage has been improved, so that messages can be forwarded properly.
* A problem with forwarding voicemails from folders other than the user’s INBOX has been fixed.
* The Zaptel channel driver can now support echo cancellers that provide 64ms or 128ms of echo cancellation per channel.

Click here to download Asterisk 1.2.15.

For more information about Asterisk Hardware, please visit the Asterisk Hardware section at VoIP Supply!

More from: Asterisk Garrett Smith

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