Recently in VoIP Category

Nimbuzz just released their new iPhone version of Nimbuzz which also supports 3G VoIP "dial up" calling and can turn the iPod touch into an iPhone. The old version was just released into the Apple iTunes store in November, so Nimbuzz is cranking out new version pretty quickly!

The new version features a full dial-pad, and the ability to make VoIP calls to PSTN numbers using SkypeOut, as well as via their 10 VoIP partners including Gizmo5, Vyke, sipgate and A1 by leveraging SIP. You can now add individual buddies from AIM, Google Talk, Windows Live Messenger (MSN), MySpace, Yahoo!, and Nimbuzz.

If Wi-Fi is unavailable you can make VoIP calls to Nimbuzz buddies using what Nimbuzz calls "Dial-Up VoIP", which is available in over 50 countries.

Dial-Up VoIP simply means that Nimbuzz dials a local access number that your iPhone dials and then Nimbuzz's VoIP servers terminate the call. Jajah, and others have this feature as well. fring also has a good app for the iPhone.

According to the Nimbuzz blog post, "We are experimenting with Twitter, and you can post to Twitter via the Personal Message feature! Try it. Your comments are always welcome, so please feel free to give feedback." Wow! Twitter integration with a VoIP app. Gotta love it!

Fixes:
•    Facebook names are displayed
•    Mobile Me usernames with a dot are now supported
•    Improved stability

Today, Dotcom-Monitor announced a new SIP monitoring tool to add to its portfolio of external monitoring services. It's similar to other web-based Monitoring-as-a-Service (MaaS) services which monitor the uptime of web servers and notify when a problem occurs. In this case, Dotcom-Monitor's SIP Monitoring service monitors on-premise or hosted IP-PBXs.

How's it work? Dotcom-Monitor's SIP monitoring service makes live intermittent SIP-based calls to VoIP devices, providing real-time monitoring, alerts, and performance reports regarding SIP component connectivity. When a problem is detected, the SIP monitoring notification feature sends an alert via phone, pager, email, or SMS. Basically ,it acts as a SIP end client, placing an actual telephone call to a specified number, and checking the results of that call.  The expected result of the call is setup as "Answer", "No Answer", "Busy", or an Error Condition (if there is an unexpected result).

According to their representative, "real-time connectivity status reports are provided via an intuitive online Dashboard interface offering sufficient detail to help pinpoint where the error condition is occurring. This reporting functionality also includes detailed historical reports and charts for managing VoIP systems and components, including Service Level Agreement (SLA) compliance issues."

I'm going to talk to then next week to find out more. For now, check out the news release...

Dotcom-Monitor Enhances Unified Suite of Monitoring Services with SIP Monitoring for VoIP Systems
Easy-to-Use, Cost-Effective External Service Monitors and Analyzes SIP Systems or Infrastructure for Uptime and Performance

Minneapolis, Minn. − March 18, 2009 − Dotcom-Monitor, (www.Dotcom-Monitor.com), a leading provider of externally-hosted network monitoring services, today announced the addition of a cost-saving SIP monitoring service to the company's unified suite of monitoring capabilities. Today's announcement adds another critical tool to Dotcom-Monitor's portfolio of external monitoring services, which includes uptime and performance monitoring of websites, web applications, and Internet network infrastructure.

Dotcom-Monitor's new SIP monitoring service makes live periodic SIP-based calls to VoIP devices, providing real-time monitoring, alerts, and performance reports regarding SIP component connectivity. When a problem is detected, the SIP monitoring notification feature sends an alert via phone, pager, email, or SMS. Additionally, real-time connectivity status reports are provided via an intuitive online Dashboard interface offering sufficient detail to help pinpoint where the error condition is occurring. This reporting functionality also includes detailed historical reports and charts for managing VoIP systems and components, including Service Level Agreement (SLA) compliance issues.

"Due to SLA requirements and hybrid VoIP traffic routes, it is important for VoIP monitoring to proactively mimic the end-user's perspective from external locations, rather than only relying on passive internal network analysis systems," said Vadim Mazo, founder and chief technical officer of Dotcom-Monitor. "Many organizations' VoIP monitoring and uptime needs are best addressed by a simple, cost-effective external system, rather than a large, expensive in-house system. Dotcom-Monitor's SIP monitoring service provides customers a unique, easy-to-use, targeted solution for quickly identifying and pinpointing VoIP connectivity error conditions," noted Mazo.

The new SIP monitoring service can be configured and managed with little or no IT expertise, which is ideal for the growing number of small and mid-sized businesses (SMBs) with on-premise or hosted IP-PBXs. Its proactive monitoring ensures connectivity errors can be addressed before the errors become downtime problems for customers. Dotcom-Monitor's SIP monitoring service ensures SMBs can rely on their VoIP systems, Service Providers can monitor their VoIP infrastructure, VoIP Wholesalers can monitor Service Provider connectivity and reliability, and VoIP VARs and managed service providers can count on client uptime and revenue.

"As the VoIP ecosystem continues to grow in scope and complexity the need for simple and affordable SIP monitoring has never been greater," said Jonathan Fuld, CTO of SIP Print, the only provider of pure, affordable SIP call recording systems for SMBs. "In fact, SMBS and any cost-conscious organization that is dependent on SIP-based communications could benefit by investigating an externally hosted SIP monitoring provider like Dotcom-Monitor."

Dotcom-Monitor's SIP Monitoring is available immediately by visiting: www.dotcom-monitor.com

Happy St. Patrick's Day! And in the spirit of St. Patty's Day, the folks at Skype have a special treat for you Mac fans. Sorry, PC users, you're out of luck this St. Patty's Day...

Skype's Peter Parkes wrote:
As a special treat, we're giving away 10,000 minutes of Skype Access completely free - simply connect as normal for up to 30 minutes at a time, and you won't pay a thing.

For those of you who are in the dark about Skype Access, here are the basics:

Skype Access is a super-cool feature only found (at the moment) in Skype 2.8 Beta for Mac. It lets you pay by the minute for WiFi using your Skype Credit - except, of course, that until the 10,000 minutes run out, you won't actually pay at all.

To get started, make sure you're running Skype 2.8 Beta for Mac, find a compatible hotspot, and surf away

Asterisk has just added 'official' MFC/R2 support for chan_dahdi. Here's the commit. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva.

According to the commit:

Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others.

To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample.

Via Russell Bryant

Office Communications Server 2007 Mediation Server uses a plus sign (+) to prefix E.164 numbers in the Request Uniform Resource Identifier (URI) for outgoing calls. Unfortunately, some IP-PBXs don't comply with RFC 3966 and do not accept numbers that are prefixed with a plus sign (+).

As the UCSpotting blog points out:

To make sure that OCS 2007 operates correctly with non-RFC 3966-compliant PBXs, Microsoft released an update for Mediation Server (R1), which is described in KB articles 952780 and 952785. After installing the update, it's necessary to create a configuration file - MediationServerSvc.exe.config - with the following content:

In OCS 2007 R2, Microsoft changed this slightly negating the need for the above configuration file. There's a new WMI setting, RemovePlusFromRequestURI, which is described in this TechNet article called Enterprise Voice Server-Side Components.

According to the TechNet article, Office Communications Server 2007 R2 introduces two new Windows Management Instrumentation (WMI) settings for Mediation Server. The first new setting specifies how Mediation Server processes E.164 numbers in outbound calls. The second new setting enables Quality of Service (QoS) marking on Mediation Server.

Handling E.164 Numbers in Outbound Calls (OCS 2007 R2)

By default, E.164 numbers in the Request Uniform Resource Identifier (URI) for outgoing calls are prefixed with a plus sign (+). Most Private Branch eXchanges (PBXs) process such numbers without problem. Certain PBXs, however, do not accept numbers that are prefixed with a plus sign.

To ensure interoperability with these PBXs, Mediation Server has a new WMI Boolean setting called RemovePlusFromRequestURI, which has two values: TRUE and FALSE. If your PBX does not accept numbers prefixed with a plus sign, the value for the WMI setting should be set to TRUE, which causes Mediation Server to strip the plus sign from a Request URI for outbound calls. The default is FALSE, which causes Mediation Server to pass the outgoing INVITE's Request URI, To URI, and From URI unchanged.

The TechNet article also discusses compatibility with PBXs that do not support the plus (+) sign.

By default, E.164 numbers in the Request URI of outgoing calls from Office Communications Server 2007 R2 are prefixed with a plus sign. Most PBXs process such numbers without problem. Some PBXs, however, do not accept numbers that are prefixed with a plus sign and do not route those calls correctly.

Additionally, the From headers of inbound calls from some PBXs does not conform to RFC 3966 because they are not prefixed with a plus sign. Microsoft Office Communicator cannot resolve these numbers to the correct user.

To assure interoperability with these PBXs, Office Communications Server 2007 R2 has a new Mediation Server setting for WMI called RemovePlusFromRequestURI. This setting can be set to YES or NO. The default value is NO.

- If a PBX downstream from the Office Communications Server 2007 R2 Mediation server does not accept numbers prefixed with a plus sign, set the value of RemovePlusFromRequestURI to YES. This causes Mediation Server to remove the plus signs from the Request URIs of outgoing calls. It also causes the plus signs to be removed from the To and From URIs.
- If the downstream PBX accepts numbers prefixed with plus signs, leave the value of RemovePlusFromRequestURI set to its default value of NO. This causes Office Communications Server 2007 Mediation Server to pass Request URIs, To URIs, and From URIs unchanged (that is, with plus signs).

UCSpotting's excellent article explains all this, and includes a nice VBscript for how to change the boolean value (true or false). Check it out!

Well, Microsoft has let the cat out of the bag and leaked word that Microsoft OCS 2010 will "remove the need for PBX equipment within your organization". I'm certainly not surprised. Let's flash back to last year where I wrote and article titled Microsoft OCS 2007 R2 Heralds the Death of the IP-PBX. In it I wrote:

"Office Communications Server 2007 R2, debuting just one year after the Microsoft unified communications launch, highlights the pace of innovation that is possible with software," said Stephen Elop, president of the Microsoft Business Division at Microsoft. "This new release puts Microsoft on a rapid path to deliver voice software that does much more than a network private branch exchange (PBX) and with much less cost."

Interesting quote, eh? Does this not sound like Microsoft is sounding the death knell for the network PBX (IP-PBX)? This is an interesting turn of events. Microsoft hasn't been pitching OCS 2007 as an IP-PBX replacement, but rather as something complementary. In fact, I remember talking with Microsoft about this last year (2007) and they went out of their way to explain that OCS 2007 is not an IP-PBX replacement. Also, Microsoft has many IP-PBX partners in the OCS 2007 arena, including Mitel, Nortel, and others. Slip of the tongue? Or is Microsoft going full-out into the IP-PBX arena? Certainly, the fear by many IP-PBX vendors is that one day Microsoft will offer a full-fledged software-based IP-PBX replacement, but I don't think that day has come yet - even with the new features in OCS 2007 R2.

Now with OCS 2007 R2 fully launched and with added support for direct SIP trunking, the next logical step is a 100% Microsoft UC solution without the need for a PBX/IP-PBX at all. Of course, Microsoft OCS 2007 R2 is still currently very limited in the support it has for SIP IP phones. Most businesses aren't ready to toss desktop phones for a 100% software-based softphone solution, i.e. Microsoft Communicator. So OCS 2010 will have to support SIP phones from popular SIP phone players such as Aastra, Polycom, and snom. Perhaps Microsoft will borrow or acquire the technology from SmartSIP, which recently launched an add-on for OCS 2007 R2 that enables any SIP phone to work with OCS.

So where did I hear that Microsoft was aiming to eliminate the need for a PBX in OCS? I discovered the information within a document on Microsoft's website titled 'Microsoft Unified Communications Business Value Tool'. On Page 24 it states:

You will deploy Office Communications Server 2010, which expands on the communications capabilities delivered in OCS 2007 R2. This release is designed to remove the need for PBX equipment within your organization and replace it with an integrated communications system that dramatically reduces management costs and gives end users innovative tools to communicate and collaborate across geographic boundaries from their office, home or on the road.

Not only do they state they will eliminate the PBX, but they declare the next version name of OCS (OCS 2010), which as far as I know Microsoft hadn't announced yet. Many UC/VoIP experts predicted that eventually Microsoft would attack the IP-PBX space alone, but one has to wonder if alienating their IP-PBX partners is such a good idea. One of their strongest OCS partners is Nortel, who is experiencing financial difficulties and is probably not in a position to pressure Microsoft to back off. Mitel is another strong partner as well that could be impacted by Microsoft's decision. Of course, Nortel and Mitel could still go after the SIP-based IP phone space within the OCS arena, but the IP phone market is much more of a commodity with a much lower margin than a full-fledged IP-PBX. Of course, there's always the high-end media phone market with large margins. For instance, Polycom recently announced their VVX1500 media phone, which created some buzz.

I doubt OCS 2010 will have all the advanced call center functionality you get from Nortel, Avaya, Mitel, etc. After all, this will be Microsoft's first release that doesn't rely on the IP-PBX to do the intelligent call routing & handling. They'll probably have some rudimentary call queues and skills-based routing, but not much else. Don't expect predictive dialing in OCS 2010, a mainstay of the call center market. Still, a 100% software-based IP-PBX with unified communications capabilities will be a compelling choice for many businesses.

A new mobile-phone MVNO carrier, Zer01 Mobile announced that it will give you 100% unlimited voice and data on smart phones for as little as $69.95/month, without a contract, and on a broad nationwide network - namely AT&T's network.

Update: (they're not a MVNO as I originally reported due to information I read on the web)

Unlike MVNO's which have agreements with a carrier to resale the carrier's service, ZER01 has no such agreement in place with any carrier, and in fact, actually provide their own voice and data to you.

Zer01 Mobile will even offer unlimited international calling (to 40 countries) for an additional $10/month. How are they able to offer such inexpensive calling? Take one guess. Yep, they're using VoIP. The company will use a VoIP application for routing of all calls. Unfortunately, currently that app only works on Windows Mobile - sorry Apple iPhone fans (& Blackberry).

Update - However, Zer01 has plans to make this service available for users of Blackberry, Palm, Symbian, and other hand set devices in the near future.

According to PC Magazine, "Zero01 has some sort of quality-of-service mojo that lets VOIP run even over slow EDGE and GPRS networks."

Zer01 works with unlocked Windows Mobile phones that can run their application, but Zer01 also plan to sell three HTC phones, the TyTnII (AT&T calls it the Tilt), the Touch Diamond and the Touch 3G.

It'll be interesting to see if people are willing to try 100% VoIP mobile phones. Will they trust the voice quality and coverage area? Then again, it is AT&T's nation-wide coverage area, and cellular voice quality isn't that great either. So users might be willing to take a chance. Certainly the price is very attractive, and with no contract requirement, customers might be willing to give it a shot.

More...

Terminal Services allows you to remotely run applications as well as perform remote administrative duties on servers. It has allowed remote audio to be streamed over IP from the remote computer to your local computer (audio redirection) but has never allowed the microphone or line-in port to be redirected. If Microsoft did, you could do VoIP. Of course, you'd have to redirect from the local PC to the remote server and not the other way around. Well read on...

AOL is dropping VoIP yet again. I'm getting dizzy from all the times they've entered the VoIP space, exited, and then re-entered.

First a history lesson on their starts & stops in VoIP/telephony.

In 2006, AOL gives up on Total Talk and throws in the towel on Total Talk, a broadband Vonage-like service (ATA), supposedly in favor of PhoneLine a 100% software-based (softphone) VoIP solution that supports both outbound and inbound (DID number) dialing.
In 2007, AIM Call Out launches offering SkypeOut like features from the popular AIM IM client. AOL has once again entered the VoIP arena. This time it appears they're offering outbound calls to PSTN number only. I see no mention of inbound PSTN calls that AIM Phoneline supports.
In 2008, AOL launches Open Voice API

And now this blog post today (2009) from VoIPGuides:

AIM Call out VOIP Service by AOL is in the process of a shutdown. Unfortunately, They never took off with their VOIP Service,it was used only by a small number of people. I understand the pain to run a costly VOIP company and having a bunch of VOIP developers sitting in your office with no profit coming from the service.

So it would appears AOL is once again cold on VoIP. I tried to confirm this by downloading AIM myself, but inexplicably their download page only lists Win 98 as a supported Windows option (See image below). They don't list Windows XP or Vista. Obviously, if AOL can't hire webmasters and designers to update their website regularly, then obviously they have financial problems. I wouldn't be surprised to see AIM die in the next year or so. I don't know anyone that uses it anymore. Skype does IM just as well, has a huge community, and it's got VoIP!

Today, Polycom has launched the Polycom VVX 1500 touch-screen business media phone, a new VoIP phone that combines IP telephony with business-class video and the ability to integrate with business applications. Recently, Verizon make a big splash with their consumer-class Verizon Hub, a multimedia phone that combines VoIP, Internet access, color screen, video streaming, and more. One could easily make the case that the Polycom VVX 1500 is the "business-class" version of the consumer-oriented Verizon Hub phone.

Although there are many similar features and both could be classified as "media phones", the Verizon Hub does not do video conferencing, since it does not have an embedded camera. The Polycom VVX 1500 on the other hand does have a video camera embedded (2-megapixel) and is therefore more suited to video conferencing, which is more prevalent in the business world any way.

The Polycom VVX 1500 combines a personal video conferencing system with a fully featured voice over IP (VoIP) telephone along with Polycom HD Voice (wideband telephony) and an open application programming interface (API) and microbrowser for real-time delivery of personalized Web content. It also includes a 7" color touch-screen interface making this a very unique business IP phone.

So is a business-class media phone with a color touch-screen, web browsing, and video conferencing capabilities a game changer in the VoIP space? Well, the VVX 1500 has a list price of U.S. $1,099, so this is not an IP phone for everyone's desk in a corporate office. A decent IP phone for the every day worker can be had for $150-$300 which is much less expensive. However, for business executives, CEOs, VPs, and other high-level management, the VVX 1500 is a very attractive IP phone. Often times if a VP or CEO has to have a high-quality video conference, they have to reserve a high-quality video conferencing system located in a particular boardroom. With the VVX 1500 they can stay at their desk and have their meeting. Further, impromptu video conferencing with co-workers sporting a VVX 1500 on their desk can be had allowing for quick collaborative meetings.

In-Stat is very high on the prospects for business-class media phones. According to Keith Nissen, principal analyst at In-Stat, "We anticipate that within five years, nearly 10 million business media phones will be shipped worldwide, generating more than U.S. $3 billion in annual revenue. They are a key to the future of the IP PBX business." He added, "With its rich heritage in voice and visual communications and content sharing, Polycom is well positioned to be a leader in this new world of communications. The company's VVX 1500 is the first business media phone that enables customers to work more efficiently and effectively than ever before by tying together voice and visual communication with critical business processes."

                                                       Polycom VVX 1500 Touch-Screen

"There is growing demand from our service providers and customers to help them configure video within our BroadWorks call control platform," said Mike Tessler, CEO of BroadSoft. "We have a long history of teaming with Polycom to deliver high quality hosted VoIP solutions, and the VVX 1500 is especially compelling because it goes far beyond the functionality of a traditional video phone by combining rich telephony, business-class video and an applications platform that is all deeply integrated with the BroadWorks platform, and it is extremely easy-to-use."

The VVX 1500 was also specifically designed for lower power consumption, using power over Ethernet (PoE) using IEEE 802.3af, and requiring less than half the power of similar competing products such as traditional video phones. The device's cool smart-motion technology enables the screen to go into power-save mode when no one is in the office.

The VVX 1500 features an open API and microbrowser that enable third-party application developers to integrate VVX 1500 with business applications such as unified communications, customer relationship management (CRM), and appointment management systems. The always-on, touch-screen user interface of the VVX 1500 includes a menu screen on which developers can place icons for users to locate and start their applications.

                        Polycom VVX 1500 Profile View

The VVX 1500 comes bundled with several applications including the Polycom Productivity Suite, which enables users to initiate and control audio conference calls right from the device's screen as well as record calls locally using a flash drive in the phone's USB port. The VVX 1500 also features a free Web service called My Info Portal through which customers can select to receive content such as local weather reports and other personalized information on the screen when the device is not in a voice or video call.

Interoperability is not a problem since the VVX 1500 uses the same Session Initiation Protocol (SIP) software as incorporated in Polycom's SoundPoint IP and SoundStation IP desktop and conference phone product lines to communicate with SIP based IP-PBXs and hosted SIP servers. The product is in the process of being SIP video-certified by Polycom's ecosystem of more than 30 VIP and VoIP Field Verified call control partners, including BroadSoft, Deltapath, NEC Sphere, Objectworld, and Zultys.

"Our customers consistently seek better leverage of their communication systems to improve productivity and reduce costs. They also expect Polycom to continuously deliver innovative, intuitive products to market," said Sunil Bhalla, senior vice president and general manager of Voice Communications Solutions at Polycom. "Our leadership and legacy in both voice and video communications enables us to develop a truly unique device. The VVX 1500 is the business media phone to combine a superior business-grade VoIP telephone that features our renowned HD Voice with one-touch video and access to key enterprise applications. We're delighted propel collaborative communications to the next level with this ground-breaking device."

The Polycom VVX 1500 will be available this month through Polycom's channel partner network at a list price of U.S. $1,099. To learn more about the Polycom VVX 1500, visit www.polycom.com/vvx1500.