SIP
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GL Communications has enhanced its VoIP stress testing tool called PacketGen, a software based VoIP testing tool that features real-time VoIP bulk call generation (including SIP signalling & RTP generation), stress testing, and analysis of VoIP networks and equipment. The latest version, PacketGen 3.0 supports SIP call generation (individual calls as well as bulk calling) and RTP/RTCP traffic generation. With PacketGen you can send/receive voice files, user-defined digits and tones, as well as create noise impairment and real-time voice traffic. TMC Labs has tested their testing products over the years, including their flagship product, the
(Digital Central Office Switch Simulator (DCOSS), which we reviewed
here and
here.
Using PacketGen, calls can also be made to IP phones and to Analog Telephone Adapters (ATAs). PacketGen can be used to test basic functionality and verify proper protocol implementation in SIP based equipment such as SIP phones, as well as Proxy Servers, SIP Registrar servers, as well as PSTN and Media Gateways. One of my favorite features is its integrated impairment functionality. You can generate automatic impairments over the RTP for any (or all) established calls. These impairments include latency (Fixed, Uniform, Nominal), packet loss (Periodic, Random, Burst with burst probability and burst size), and packet effects (Out of order, Duplicate Packets).
Here is a list of their new features:
Continue reading PacketGen VoIP testing tool...
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Written by VoIP & Gadgets Blog on September 28th, 2006 with no comments.
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Rich just wrote about an interesting conference session titled
SIP Trunking - Realizing Rapid ROI Today, taking place at Internet Telephony Conference & Expo (ITEXPO) in just 2 weeks. Although I
ditched my residential Vonage VoIP service, due to price and QoS issues, that doesn't mean I don't whole-heartedly support SIP trunking in the corporate enterprise -- where voice is even more critical. Most corporations have better QoS Internet/IP connectivity than residential Internet offerings, thus SIP trunking with "guaranteed QoS" is a viable option to an expensive voice PRI/T1/E1 line. Some service providers offer voice VPNs for added security. If your attending ITEXPO, definitely check out this session to see how SIP trunking can save on telecom costs.
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Written by VoIP & Gadgets Blog on September 26th, 2006 with no comments.
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ADTRAN today announced the introduction of the NetVanta 3400 Series of Multiservice Access Routers. The series delivers up to two T1s of wire-speed performance, even with advanced services like Firewall, Access Control Lists (ACLs) and IPSec Virtual Private Networking (VPN) enabled. I've always been a fan of ADTRAN's "all in one" networking solutions, which often give you not just a router, but PoE, firewall, VPN, built-in SIP server, CompactFlash for backup, and more. Back in May,
I wrote about ADTRAN's 7100 IP-PBX, which I thought was one of the most comprehensive, plug-in-play, all-in-one IP-PBXs in the market. Similarly, I was
impressed with ADTRAN's Total Access 900, which is also a comprehensive solution.
Well, the new NetVanta 3400 series includes the NetVanta 3448 and 3430, each a modular, 1U-high, rackmountable metal chassis that also follow Adtran's "all in one" model. These products are targeted at the Small and Medium Business (SMB) or enterprise applications requiring high-performance throughput for bandwidth-intensive applications. Each offers a single-slot to house any of the NetVanta Network Interface Modules (NIMs) for WAN access and Dial Backup Interface Modules (DIMs) for disaster recovery applications, two 10/100Base-T Ethernet Local Area Network (LAN) interfaces that allow for broadband backup or DMZ, CompactFlash for configuration backup and storage of multiple firmware revisions, and support for up to 500 simultaneous VPN tunnels. Each is RoHS compliant making it ideal for both domestic and international applications.
The NetVanta 3448 includes the addition of a fully managed, non-blocking, eight-port 802.3af-compliant Power over Ethernet (PoE) Switch delivering a full 15.4 watts of power per port. This allows you to power any PoE device, including security cameras, SIP phones, or wireless access points.
These products also offer a five-year warranty and free firmware upgrades for the life of the unit.
“As the demand for bandwidth-intensive applications continues to grow so does the need for greater throughput and processing speed,” said Rick Schansman, senior vice president and general manager, ADTRAN Enterprise Networks Division. “The NetVanta 3400 Series clearly addresses these needs by allowing businesses to fully utilize a complete arsenal of security and performance features without degrading throughput performance.”
The NetVanta 3430 has a list price of $895, while the NetVanta 3448 has a list price of $1,045. Both models are currently shipping. The Power over Ethernet option for the NetVanta 3448 has a list price of $345 and will be available in Q4. Enhanced VPN capabilities can be activated through a software upgrade for $395.
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Written by VoIP & Gadgets Blog on September 19th, 2006 with no comments.
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Note: Nice little article explaining SIP in a basic sense. Informative Read
Have you ever wondered why long distance calls cost so much? In part the reason is because telephone lines cost so much. When driving, you might occasionally see a telephone crew maintaining a telephone line, but what you may have never considered is that there are literally thousands of individuals working around the clock to maintain our telephone lines.
The telephony system works via a cog and wheel setup. What this means is that every long distance call you make is routed along a telephone wire to a central station, where your voice is routed to another central station, which is finally carried to the person with whom you are trying to communicate. For the call to be maintained, the entire time you are speaking, a space along all the lines in between you and the person you are talking with must be completely devoted to you. Because millions of people are talking at the same time, the little space along the telephone lines becomes rather desired property. And like all things desired, the price is high. Before recent innovations, however, there were no alternatives, so everyone grudgingly paid the often costly long-distance telephone bill.
SIP, or Session Initiation Protocol, has turned the telephony world upside down. Specifically, SIP refers to a protocol that allows computers to talk to each other without going through a central station. Practically, what that means for you and me is that it is no longer necessary to pay for expensive telephone lines to complete our calls. SIP technology is a relatively new development in which calls are made on a peer-to-peer rather than cog and wheel network. What that means, is that you are now able to call people directly from your SIP enabled phone to theirs. This ends up being radically cheaper than the old way of calling.
The SIP system does not require a central computer and operators like the old telephony system did. Rather, your computer, or SIP enabled phone, does all the routing for you.
SIP has been around for a number of years, but only recently has it begun to go mainstream and take off in popularity. This quick increase in interest over SIP is due to companies like Mobalex, who were aware of the fact that over the generations we have come to expect certain tones, buttons, and protocols from our phones. So what they have done is to transpose those functions onto the SIP system. Rather than forcing users to communicate in a completely new way, what these companies have done is to provide a calling experience which from the user’s perspective is completely identical to traditional telephony.
SIP is typically offered in two formats, computer based and hardware based. Computer based SIP is a system that allows you to make calls using your computer as the router and communicating via a headset on your computer. The more practical and popular version, however, actually provides you with new SIP enabled telephone handsets or converts your existing phones to SIP. By eliminating any technical requirements, modern SIP providers have made using the system as easy, or easier, than using a traditional phone. I say easier, because many companies are able to take advantage of the fact that the system is internet based to provide you with some very unique benefits. These include the ability to adjust your plan, change your calling options, and even pay your bill from the same website.
SIP technology is quite revolutionary in the world of communication. By creating a peer-to-peer network, SIP has been able to radically undercut the prices of traditional telephony, take advantage of the Internet, and still maintain the ease of traditional telephony. It is merely a matter of time before we are all using SIP for all of our telephoning needs.
Source: Click Here
Written by Dal on September 13th, 2006 with no comments.
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Truphone today announced availability of a free beta VoIP software download for Wi-Fi-enabled, mass market mobile handsets. The first download is available for Nokia's latest E-series handsets, with the N-series handsets to follow imminently. Support for additional Wi-Fi-enabled phones, including those built on the Windows Mobile platform will soon follow.
Calls between 'on net' Truphone handsets are free worldwide. However, in the footsteps of Skype's
free U.S./Canada SkypeOut promotion (amongst others), Truphone has a launch offer to the end of the year (Dec 31st 2006) that gives
USA users free calls to USA/Canada and UK users will be able to call some
two billion landlines in major countries for free. All other Truphone calls from the user's mobile will be charged at low VoIP rates. (e.g. UK 2.7c/min, China 2.8c, Mexico City 3.1c, Japan 4.8c)
Provisioning, installation and use are a breeze. Users send an SMS with the word 'TRU' to a designated number, provisioning takes place 'over the air' and the software installs itself quickly and easily on the mobile handset. Truphone integrates seamlessly with the phone using the existing address book, so there's nothing to re-type. Once set up to use a Wi-Fi hotspot, the handset connects automatically next time the phone is in range.
Dialing is a breeze using this software since it is seemlessly integrated Simply dial or select a contact as you normally would and press the Call (Green) button. Call behaviour is identical to using your handset over your usual GSM/Cell provider. The handset will attempt to make a Truphone call before trying a GSM/Cellular call. If you want to manually select the call type, dial or select a contact as you normally would and select
Options>Call>* Call where * = Voice, Video or Internet.Support for Truphone SMS messaging is imminent, while presence functionality will also follow. The SIP-based software incorporates several leading-edge techniques, including the self-healing, small-footprint AMR codec to ensure optimal call quality even under less than optimal conditions; and, unlike other Wi-Fi applications, Truphone uses smart, low-level coding to minimise battery usage.
James Tagg, Truphone's chief executive, said: "Truphone is the natural evolution of VoIP. First people talked to a PC, then they were tied to fixed-line handsets and now Truphone takes VoIP mobile. Free calling has arrived on mass-market handsets."
To get Truphone:- Users in the UK should send an SMS with the word TRU to shortcode 60030
- Users in the USA should send an SMS with the word TRU to +44 7624 000 000
- Users elsewhere should send an SMS with the word TRU to +44 7624 000 000
Truephone is certainly leveraging some good open-source stuff, as well as several popular Linux components.
The major telecoms components are:- OpenSER: Open source version of the SIP Express Router
- Asterisk: Open source SIP PBX
Other components that they use are:- Apache Tomcat Servlet Container
- MySQL Databases
- Debian Linux Build
- Spring, Hibernate, JSF, Shale - Clay, Acegi for the web site build
- Roller for blogging
- Nagios and Monit for status monitoring
- OTRS for trouble-ticketing
- Jira for bug tracking
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Written by VoIP & Gadgets Blog on September 12th, 2006 with no comments.
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Flextronics Software Systems (FSS), the global leader in communications software, today announced that it is the worldwide share leader in the Session Initiation Protocol (SIP) stack market, according to a recent Venture Development Corporation (VDC) report.
FSS is the clear leader in the SIP stack market with 32% share, according to VDC’s Telecom Core Infrastructure Market Intelligence Service 2005-2006. SIP is an application layer control protocol that can establish, modify and terminate multimedia sessions or calls.
“SIP is a dynamic and rapidly growing software sector where scalability and interoperability are critical,” said Robert Johnson, Senior Telecom Analyst of Venture Development Corporation. “Hardware manufacturers, software vendors and service providers have increasingly high expectations of companies providing SIP protocol stacks and toolkits. FSS’ top ranking is a very strong statement from the market,” continued Mr. Johnson.
“FSS’ products’ proven ability to reduce time-to-market and their superior scalability, interoperability and quality are the top factors our customers cite when they choose our SIP products,” according to Suresh Kabra, Assistant Vice President and Head of Products of Flextronics Software Systems. “We will continue to accelerate the development of our customers’ products and services through FSS products that include SIP-powered IMS offerings, SIP User Agent Toolkits, our SIP Server Framework, our Back-to-Back User Agent and our SIGCOMP stack,” added Mr. Kabra.
FSS protocol stacks, frameworks and toolkits have been critical to the delivery of products from over 200 Original Equipment Manufacturers (OEMs) worldwide. Flextronics Software Systems offers innovative software products for:
-- wireless, wireline, broadband and satellite infrastructures;
-- carrier-grade and enterprise class communications equipment; and
-- devices, server-side communications applications and the network core.
Source: BWI
Written by Dal on August 30th, 2006 with no comments.
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Palm users have been clamoring for VoIP on their Palm devices for quite some time. I've mentioned VoIP on Palm (
including the Treo) in the past.
mobiVoIP was the first to offer VoIP on a Palm device. However, mobiVoIP only works with their VoIP service plan. Many Palm users simply want a SIP softphone client that works with
any SIP-based termination service provider. I also
wrote about Talkplus, but they don't support SIP either and they too use their own phone service plan.
Well, today,
Hampton Software Limited announced the release of a new Palm VoIP product called Articulation for PalmOS, that supports the SIP standard and therefore works with any VoIP service provider that supports the SIP standard. I should mention that it only works with Palm OS 5. Alas, the software doesn't yet work on the Treo 700p mobile phone, but should work on the Treo 650. It is connectivity-agnostic supporting WiFi, Bluetooth, EVDO, and EDGE/GPRS Internet connectivity.
Screenshot of Articulation for PalmOS
It isn't the prettiest interface in the world, but hey, if I can get my SIP on, I'm a happy man!
Features:- Ability to make PSTN calls with your PDA
- Select the SIP VoIP provider of your choice
- Lookup phone numbers direct from you contacts
- Touch-tones (DTMF) for 'phone menus'
- Supports WiFi, Bluetooth and EVDO connections
- Call timer
- Silence suppression (only transmit your voice)
- Echo cancellation for 'speakerphone' use
Technical Features:- Secure account authentication with SIP MD5 Authentication
- Supports GSM and G711 codecs (GSM is suited to low bandwidth connections)
- NAT support through RFC 3581 and ability to fix ports for symmetric NATs
- Symmetric RTP support
- Low latency
- Low memory usage
Good stuff! You can
download and try it out for free. Still wish it supported the Treo 700p though.
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Written by VoIP & Gadgets Blog on August 23rd, 2006 with no comments.
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Ingate Systems, which develops firewall technology and products that enable SIP-based live communication for the enterprise while maintaining control and security at the network edge, showcases traversal and security solutions for enterprises connecting to SIP trunks, with Ingate Firewal and Ingate SIParator products at the VoiceCon show.
SIP trunking is a service offered by Internet Service Providers (ISPs) which permits businesses to adopt Voice-over-IP (VoIP) with its attendant benefits and remain connected to others who rely on the PSTN. For the enterprise SIP trunks can offer significant cost savings, as they eliminate the need to purchase either the local PSTN gateway or costly ISDN BRIs (Basic Rate Interfaces) or PRIs (Primary Rate Interfaces). The local IP-PBX can be connected to the service provider's PSTN gateways over the Internet.
Ingate Firewalls and SIParators solve the Network Address Translation (NAT) traversal issues that are faced by businesses using a SIP trunk. All voice traffic (as well as data traffic) must traverse the enterprise firewall/NAT. However, SIP traffic cannot traverse traditional enterprise firewalls and NAT devices. This is because traditional firewalls do not differentiate between SIP traffic and unwanted traffic. As a result, the firewall/NAT device blocks all SIP traffic, which includes VoIP. Ingate products resolve this issue, enabling enterprises to utilize SIP trunks.
Ingate's fully SIP-capable, proxy-based Firewalls and SIParators also provide flexibility to interoperate with carrier-specific requirements like numbering plans and authentication. They establish least cost routing rules, which offers businesses the ability to use multiple service providers, and to switch between them depending on which offers the best possible rates (which may vary by time, day or location). Long distance calls cost the same as a local call, reducing expenses for businesses as well as their customers, partners, etc. trying to reach, for example, the corporate sales force.
SIP trunks further reduce costs as they eliminate the need for separate voice and data connections, expanding the potential for communications convergence using both voice and data together. SIP trunking also offers scalability so that, as a company grows, all necessary infrastructure to handle additional voice/data traffic is already in place.
Security Over The Public Internet
Ingate products are specifically designed to leverage all the security benefits available with SIP communications. Ingate's enterprise-class Firewalls secure data and SIP traffic, while SIParators secure SIP media while leaving the traditional firewall in place (working in parallel to the SIParator) to secure data traffic.
Both products feature Ingate's full SIP proxy technology, which allows for advanced filtering, verification, authentication and routing, as well as dynamic control of the opening and closing of media ports. Encryption of the signaling is done using Transport Layer Security (TLS) and of the media (voice, video, etc.) using Secure RTP (SRTP). With encryption, the sessions are kept private with no chance of eavesdropping.
They also secure full VoIP redundancy, as traffic can be routed to a back-up carrier if the primary carrier is unavailable. And Ingate's proxy-based solutions can be used to provide local call management if the service provider cannot be reached; and, if the enterprise has a local gateway installed, outgoing calls can be routed to the PSTN if all connections to the hosting provider are unavailable.
"SIP trunking offers enterprises the benefits of converged communications and saves substantial expense by eliminating the need to purchase their own PSTN gateway," said Steven Johnson, President, Ingate Systems. "NAT/firewall traversal is a critical issue in any SIP trunk deployment. Ingate's SIP-capable Firewalls and SIParators offer an elegant solution to this problem, while also securing voice traffic."
Connecting Remote Workers To The SIP Trunk Without VPN
SIP trunking, when used in conjunction with Ingate's Remote SIP Connectivity software module, allows remote users to traverse most SIP-unaware residential firewalls and NAT devices and use all the IP-PBX functions installed in the enterprise. Remote SIP Connectivity complements SIP trunks, as remote workers can use the software to interface with their corporate IP-PBX and, if they were making calls to outside the enterprise, do so using the SIP trunk.
For more information, please visit us at Ingate's VoiceCon booth #639, or online at www.ingate.com.
Written by Dal on August 22nd, 2006 with no comments.
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Bandwidth.com, a nationwide provider of complete business telecom services, today announced the launch of its SIP Trunking VoIP product. SIP Trunking is used extensively by businesses that have purchased SIP-enabled IP PBXs to replace their traditional phone system in an effort to lower-cost and benefit from more flexible voice services.
Bandwidth.com's SIP Trunking product eliminates the need for additional hardware to convert TDM traffic to VoIP, providing a simple end-to-end SIP VoIP connection to the Company's network of carrier gateways. This product is unique in that it allows a business to oversubscribe each VoIP trunk; this enables any company to purchase trunks only for the number of concurrent calls they support, rather than buying one for each individual employee.
By implementing Bandwidth.com's SIP Trunking product, users are able to extend the flexibility and cost savings of VoIP to their IP PBXs without upgrading or buying new hardware. Bandwidth.com will offer the product including unlimited incoming and local calls and very competitive long distance rates, for under $.02 cents per minute. "Bandwidth.com continues to respond to changing market dynamics and customer requests by creating a solution that reduces the cost of telecommunications, simplifies IP PBX technology and enables new forms of business communications across traditional boundaries," said Henry Kaestner, CEO of Bandwidth.com.
"Bandwidth.com recognizes that many of our customers prefer premise based systems, and that deploying a PBX can be the best option for many of them. At the same time these customers want to take advantage of the cost savings of business class VoIP, and our SIP Trunking product does just that. By offering SIP Trunking in addition to hosted voice services, we continue our goal of providing the products that enable businesses to communicate using the most advanced technology platforms available."
Written by Dal on August 8th, 2006 with no comments.
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I am not a technical expert on Asterisk, but being fully aware of its gathering momentum as an is a free software / open-source software implementation of a telephone private branch exchange (PBX), I decided to hang out with some Asterisk code jocks for the better of an afternoon. I was much looking [...]
Written by Russell Shaw on July 30th, 2006 with no comments.
Read more articles on General and Research and SIP and OSCON 2006.
There are a few general purpose VoIP call recording solutions
that I've written about. Arcosoft Inc., is the latest to announce a VoIP call recording solution released today called
VONaLink SoloRecord. SoloRecord works with any VoIP phone system based on the open SIP standard, such as Vonage, to record phone calls and to provide screen pops.
With traditional phone systems, calls are recorded with either analog equipment or expensive, proprietary products from the phone company. With the latest VoIP systems built on open, standard protocols, calls can be recorded by monitoring network packets.
VONaLink SoloRecord works with any SIP based VoIP system. Soft phones, hard phones, or analog phones via an analog telephony adapter (ATA) are supported since they are converted into IP packets which can be captured. The call is recorded as a stereo WAV or MP3. An inaudible watermark can be added to the recording for later verification that the file has not been changed.
Using the caller ID of the incoming call, SoloRecord searches for the caller in
Microsoft Outlook contacts, or launches custom applications to search the web or your company database. If the caller is found, the information is popped on the screen, thereby increasing the efficiency of the customer service agent.
VONaLink SoloRecord runs on Windows XP, 2003, and 2000. The price is $99 USD per user, available now for evaluation download from
www.vonalink.com
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Written by VoIP & Gadgets Blog on July 28th, 2006 with no comments.
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Recently, I received an email from a pizza delivery solutions provider seeking my assistance in an interesting application that involves VoIP, GPS coordinates, Bluetooth, and of course pizza. This is not the first time I have linked pizza with VoIP. No siree bob! In fact, my
Vonage VoIP line resulted in me getting a cold pizza. Almost
sued Vonage over my damn cold pizza too.
I know what you're thinking. What kind of VoIP application does a pizza delivery company need, right? Well, the best way to explain the application that ties VoIP, pizza, Bluetooth, and GPS together is to include the request they sent to me.
We sell software services to pizza chains. Our system tracks drivers as they drive around town. Since we know where they are and where they are going, when they are about 4 minutes from the customers’ door, we want our computer to initiate a telephone call to the customer to say the driver will arrive momentarily. We imagine some kind of a softphone application running on the PC connected to a VoIP access point to the phone system. Our software already has the .wav files prepared to “speak” to the customers.
Any suggestions on how to configure this or who we should be talking to that might sell the necessary hardware software? Is open source stuff available?
PS Our system is written in C#, so is pretty flexible.
Pizza Pilot recently announced the completion of a multi-store test in Boston-area Domino's Pizza Franchises. Pizza Pilot was successful in reducing labor, mileage theft and lost drivers. New tools for measuring each driver's "Smart Hustle" factor as well as data mining techniques to identify and prioritize target customers with Platinum, Gold or Silver service levels were also introduced.
Pizza Pilot is a software-based product that works with any POS-including Pulse and TMS/National Systems. In combination with Bluetooth and GPS-enabled cell phones, it tracks the location of each delivery driver every 60 seconds. Pizza Pilot mapping software determines the optimal dispatching and automatically assigns orders to drivers-allowing managers to focus on inside operations.
Due to its ability to track each driver's progress on their way to each destination, Pizza Pilot can determine the moment actual deliveries are made and to update drivers' estimated return times as they return to the shop-making subsequent dispatches more accurate. Maps or on-board navigation is also available which reduce lost drivers, while real-time tracking identifies and discourages unauthorized stops.
I'm most impressed with the fact that they want to call you when the pizza is 4 minutes away. Imagine that -- a pizza company that is courteous enough to call you when they are about to arrive. How many times have you been told the pizza will take 30 minutes only to see 90 minutes roll by; then when you call to check on your pizza they simply tell you the pizza guy is on the way. Really they have
no idea where he is and they're simply patronizing you. Well, with this pizza delivery company, they track all their vehicles so they can tell you exactly where the vehicle is and on top of that they plan to call you using an automated dialer using a VoIP line.
Now if that isn't a
hot application for VoIP, I don't know what is. Mmmm.
hot pizza... Ahhhh. GPS & Bluetooth gadgetry... Ooooo... VoIP coolness... Ahhhhh... All I need now is some cold beer &
ESPN -- and I'll be in heaven. Maybe I'll launch
SightSpeed 5.0 and watch some ESPN from work.
I suggested to Pizza Pilot that there is C# code out there to initiate SIP calls and they he may want to consider looking at Asterisk, the open-source IP-PBX. I also referred him to
Erik Lagerway and Ward Mundy over @
Nerd Vittles, an Asterisk blog. Erik suggested that since he was using C#, that something in
Microsoft LCS might do the trick - or paid kits like those offered by
Counterpath. Ward Mundy said he could code something for him and asked for more details on the size and scope of the project.
I'll keep you posted if the pizza/GPS/Bluetooth/VoIP application ever goes live.
GPS VoIP pizza delivery just might be coming to a neighborhood near you!
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Written by VoIP & Gadgets Blog on July 25th, 2006 with no comments.
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BandTel announced this morning it is the first to solve the throughput and redundancy problems on high capacity SIP-based networks with its new N-Plus architecture. The solution lifts the burden placed upon servers by creating a clustered architecture that eliminates the need for numerous IP addresses on numerous SIP proxies and eliminates any single point of failure. Of course, TMC has a N+1 RAID-5 Exchange Server, but that didn't stop 2 simultaneous hard drive crashes last week that broke the RAID array bringing down the Exchange Server. RAID-5 redundancy my %@$^&*! On a usually reliable
Dell server no less... Probably a SCSI controller failure.
In any event, BandTel's N-Plus network is based on several pairs of DNS servers that direct the SIP calls to SIP Signaling Transfer Points (STPs), which in turn direct those SIP call on one of "N" SIP proxy's in the BandTel SIP proxy matrix. You can check out the full BandTel news
here...
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Written by VoIP & Gadgets Blog on July 24th, 2006 with no comments.
Read more articles on Uncategorized and VoIP and SIP and raid and redundancy.
Paragon Wireless released the world first commercial GSM/VoWLAN dual-mode smart phone,
hipi, in March 2006 and was my pick to be a winner in this years' TMC Labs Innovation Awards. With a tri-band GSM/GPRS (Class 10) radio and an IEEE 802.11b WLAN chipset, hipi enables users to enjoy broadband multimedia services at WLAN covered homes, offices, hot spots/zones as well as reliable GSM/GPRS service anytime anywhere. Featuring a 2.4 inch TFT touch screen, QVGA, with 260k Colors, the hipi also supports STUN-based NAT traversal, the SIP standard, as well as G.711, G.729a/b, and G.723 codecs. It even has some gadgety-bling via it's built-in MP3 player and a QVGA/QCIF camera. The hipi can perform SIP-based seamless handover between GSM/VoWLAN. Importantly, it utilizes a unified phone book for both GSM and VoWLAN dialing and a unified GUI for the main applications i.e. phone, E-mail, QQ (IM), and browser.
EXCLUSIVE! The current hipi is based on Linux, but I asked about a Windows Mobile 5 version since there are many popular 3rd party applications written for Windows Mobile and they said that a Windows model was indeed in the works - with some beta trials currently and plans for release later this year. You heard it here first -
the first dual-mode Windows Mobile 5 phone out later this year!
According to Paragon, the Windows-version will have comparable battery/performance characteristics of their current Linux model.
The hipi has excellent performance in power management, mobility management, security, mobile VoIP, and voice quality. hipi has passed most of regulation certification programs and has done interoperability testing with over 40 VoIP service providers, system integrators, and infrastructure equipment vendors worldwide. Standby time
>100 Hours (GSM on, WLAN on)
>200 Hours (GSM on, WLAN off). Talk time on VoWLAN is
3.3 Hours and for GSM it's
7.8 Hours. With such excellent battery talk times and standby - even with two radios operating - hipi is an ideal device for fixed mobile convergence. Check out the specs...
Hardware Specification (Linux model)•
Intel PXA271 processor with embedded Linux
• 2.4 inch TFT touch screen, QVGA, 260k Colors
• Built-in speaker/microphone, 2.4mm stereo and headset
• 1.3M pixel CMOS camera
• USB slave
• Mini SD
• 1100 mAh Li-ion battery
GSM Specification • Frequency bands: 900/1800/1900 MHz
• GPRS Class 10
• SMS, MMS, WAP applications
• FTA/CTA certification
• FCC/CE certification
WLAN Specification • IEEE 802.11b
• RF channels: US: 11, ETSI: 13, Japan: 14
• High-gain internal antenna
• WEP 64/128 bits, WPA, 802.1x
• EAP PSK/LEAP/PEAP/TTLS/SIM
• Power saving modes
• Fast roaming between access points
VoIP Specification • SIP: IETF RFC 3261
• Codec: G.711, G.729a/b, G.723
• Acoustic echo cancellation
• Dynamic jitter buffer
• Voice activity detection
• Stun-based NAT traversal
Input Methods • Handwriting Recognition > English > Chinese > Numeric characters
• Soft Keypads > Qwerty > Standard phone dialpad > Symbol
Power Management Features • Standby time >100 Hours (GSM on, WLAN on) > 200 Hours (GSM on, WLAN off)
• Talk time > VoWLAN: 3.3 Hours > GSM: 7.8 Hours
• MP3 play time > 5.8 Hours (GSM on, WLAN on) > 6.2 Hours (GSM on, WLAN off)
Fixed Mobile Convergence Features • Simultaneously activated GSM and WLAN air interfaces
• Handling simultaneously GSM and VoWLAN incoming calls
• SIP-based seamless handover between GSM/VoWLAN
• Automatic/manual switch for out-going calls between GSM and VoWLAN
• Automatic/manual switch for data applications using GPRS or WLAN
• Unified phone book for both GSM and VoWLAN.
• Unified GUI for applications (phone, E-mail, browser, QQ)
Call Features • Call hold
• Call waiting
• Call mute
• Call forward
• Call transfer
• 3-way conference
• Voice mail
• SMS over SIP
• Phone book - (1000 entries with photos)
• Incoming call prompt with picture
• View phonebook during call
• Enter sketch pad during call
• Adjust volume during call
• Auto-answer/flip answer
• Quick silence
• Turbo dial
• Manual/Auto/Earphone redial
• Call history (20 entries) Data Application Features
• POP3 E-mail client (SSL support) > 100 full E-mails with attachments up to 200KB > Document viewer for MS-Office and PDF files
• Web browser: HTML4.01, JAVAScript1.5, SSL3.0, HTTP1.1, CSS1.0
• Instant messaging: QQ Multimedia Features
• Video format: MP4, 3GPP
• Audio format: MP3, WAV, MIDI, AMR
• Picture format: WBMP, BMP, JPEG, GIF
• Camcorder: QVGA, QCIF
• Media Player > Audio: MP3 player > Video: up to 30 frames/second QVGA MP4/3GPP PIM Features
• Calendar
• Schedule management
• Alarm clock
• Voice recorder
• World time
• Currency converter
• Anniversary Other Features
• English <-> Chinese dictionary
• Calculator
• World time
• Notepad
• Sketch pad
• File transfer
• Counter
• Timer
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CommuniGate Systems says it has demonstrated their All-Active Dynamic Cluster SIP Farm on HP hardware can be scaled up to support 10 million VoIP subscribers. Multiple vendors were involved in the emulated real-world environment, including Intel, Navtel Communications, and F5 Networks. HP provided a 64-CPU Integrity Superdome system that ran CommuniGate's SIP Farm technology.
CommuniGate is claiming that the demonstration is the first milestone in building large-scale, standards-based communities of VoIP subscribers, since traditional telecommunications, cable, and wireless providers often support over 50 million subscribers. The benchmark simulation was conducted to provie cost-effective scaling by just adding notes, using either virtualization or physical additions of servers to the running cluster as needed.
Call load generation topped out at 1,000 SIP calls per second with up to 192,000 unique registered user-agent enpoints driven by a Navitel device for inbound calling. Simultaneously, the open source "sipp" SIP load generator was used to generate additional outbound calls.
Using a virtualized cluster on a single Superdome system, the benchmark team discovered that the CommuniGate SIP Farm and F5 supplied hardware continued to perform within required QoS until no further load could be generated by the test environment available. Future tests will use a fully-equipped Navtel chassis to increase available SIP load capability. A full whitepaper report of the test demonstration is available at: http://www.communigate.com/content/whitepapers.htm
Written by Dal on July 18th, 2006 with no comments.
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Cistera Networks, the leading provider of advanced IP phone application platforms and engines in the Enterprise VoIP Telephony environment, today announced the availability of its Version 1.6 Presence Engine, PresenceManager.
Demand for PresenceManager is strong across all vertical markets, and successful deployments have already been made for large customers in the education vertical market including Corpus Christi and Katy Independent School Districts.
Presence Engines allow IP phone users and administrators to tailor their environments based on multiple factors, including physical presence and availability. The Cistera ConvergenceServer™ uses PresenceManager to appropriately deliver, record and playback calls, to present notifications and events correctly and to allow a more comprehensive real-time directory view of an organization.
"In order for customers to transition from traditional telephony to VoIP, we need to ensure the availability of features they have come to depend on. What used to be called 'Do-Not-Disturb' or 'Time-of-Day Restrictions' are actually early implementations of what is now known in the industry as 'Presence,'" said Cistera Networks CTO Greg Royal. "We took those initial features and brought them into the 21st century, while adding additional functionality that is available in a VoIP environment."
Cistera is one of only a handful of vendors that can deliver both "local" presence, where users set their own presence, and "global" presence, where the administrator can set the presence of large numbers of phones. "We have one customer that can set the presence of over 6,500 phones on a daily basis. The Cistera ConvergenceServer manages and routes over 10,000 calls a day for that customer through Cistera PresenceManager," said Royal.
"Because we natively support Session Initiated Protocol (SIP) on our platform, we can extend it to include SIMPLE (SIP for Instant Messaging and Presence Level Extensions) platforms such as the Microsoft Live Communications Server (LCS) 2007, and Cisco's Communicator products," continued Royal.
The Cistera ConvergenceServer (CCS), with its robust suite of application engines, offers customers IPT applications that are easy to integrate and install. The CCS adds critical competency and features such as text and audio broadcasting, messaging, recording and content streaming within Cisco and Nortel Converged Communications environments.
Source: Cistera
Written by Dal on July 13th, 2006 with no comments.
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Barrett Lyon has created a don't miss resource that shows you how to configure P2P SIP URI (Uniform Resource Identifier) dialing on your Asterisk-based PBX."There are thousands of people that operate their own Asterisk based PBX systems, yet they do not enable any method to allow for p2p sip URI dialing," Barrett explains. "These [...]
Written by Russell Shaw on July 12th, 2006 with no comments.
Read more articles on Vonage and General and Softphones and SIP.
With more softphones on the market than you can shake a stick at, each with their own "island" of users that cannot bridge to other islands, I've become a bit disillussioned with the VoIP softphone market. Sure, there is talk about interoperability, support for SIP/SIMPLE, XMPP support, etc. but we still don't have a single unified client that can speak to
Skype users, AOL/AIM users, ICQ,
Google Talk, MSN Messenger, Yahoo! Messenger, etc. Although, software such as GAIM is a multi-protocol client that aims to unify all of these IM/softphone clients, it too is incomplete. For one, it was designed mainly to handle instant messaging (IM) and not voice or video. Although, a branch of the open-source project called
GAIM VV was started to work on adding voice and video support, but it appears to be a dead project since the last posted news was October 2005.
This lack of a single unified client means I have to have like 4-5 IM/softphone clients running on my PC, which uses more memory, requires more time spent keeping each client up to date, uses more hard disk space, Registry bloat, and other inconveniences. It got me thinking though -
what features would I like to see to create the perfect VoIP softphone and IM client?
And so, here is my "wish list" for the perfect softphone/IM client...
Continue reading The Perfect VoIP Softphone and IM client...
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Tomorrow,
American Telecom Services, Inc. (ATS) will announce the release of its first DECT 6.0 cordless multi-handset Internet phone. The master units (E6501) are expected to retail at a sub $50 mark (likely $49.99) and the extensions (E6502) will each retail for sub $30 (likely $29.99).
The E6501 includes ATS’ patent-pending Digital Clear functionality and is expandable to up to five total handsets. ATS claims this is the first cordless phone to use DECT technology that integrates a router and a SIP VoIP platform in the charging base of the master cordless phone unit. Uniden makes a cordless phone system, which I
recently reviewed - the
UIP1869V, that supports SIP, but it's not DECT and its locked to Vonage, so the ATS phone system is the first cordless phone system (DECT or otherwise) that is an open SIP system to use with any VoIP provider that provides you with SIP credentials. Nice thing about that is if you change VoIP providers you aren't left with a "bricked" phone system.
The ATS E6501 and E6502 will be sold by American Telecom under the ATS brand bundled with service through major retailers (online and offline) in the U.S. They are not announcing which service providers and/or retailers will be carrying it at this time, however if you look closely in the photo you can see SunRocket displayed, a well-known VoIP service provider. The phones will also be available wholesale to service providers who wish to distribute them through their own channels.
Basic specs are as follows:
- DECT 6.0 communication with a range of more than 650 feet
- Expandable design up to 5 total handsets
- Integrated adapter and router
- SIP compliant VoIP platform
- LCD display
- Page/locator button on base
- Intercom between handsets
- Speakerphone on all handsets
- Headset jack on all handsets
- 60 number phone book
- Auto answer
- 10 hours of talk time
- 10 selectable ring tones
- Hearing aid compatible
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| Ratings | Score |
| Installation |  |
| Documentation | |
| Features |  |
| Usability |  |
| Value |  |
| Performance | |
| Overall | |
The
Linksys WIP330 Wireless-G IP Phone is one of the coolest gadgets I've tested in awhile.
Linksys sent me a WIP330 for a product review two weeks ago and I played around with it quite a bit, but just never found the time to write the review. Well, here goes, an exclusive look at the
Linksys WIP330 Wireless-G IP Phone...
The
Linksys WIP330 Wireless-G IP Phone enables VoIP service through a Wireless-G network connected to the Internet to make low-cost VoIP calls through your Internet Telephony Service Provider (ITSP). The handset features peer-to-peer dialing, speed dial, 3-way conferencing, call waiting, call transfer, and call forward, mute, hold and selectable ringtones. The
Linksys WIP330 measures 46.7 x 135.2 (142.0 with Ant) x 18.8 x mm (1.84 x 5.32 (5.59)x 0.74 in) and weighs 0.119 kg (4.20 oz). It uses a 3.7V 1250mAh Lithium Battery, and includes a 5V 1.0A AC Adapter, but can also be charged using a mini-USB plug.
It's worth mentioning that
Linksys makes a similar model, called the WIP300. The WIP300 and WIP330 are similar in that both models let you do VoIP from 802.11b/g WiFi networks and support SIP and the SIP v2 standards. The similarities end there. The WIP330 is the higher-end model, sporting a larger 2.2" LCD screen vs. 1.8" LCD for the WIP300. The WIP330 unit adds hotspot support and the ability to browse the Web on its 2.2-inch color display. The WIP300 on the other hand doesn't have Web capabilities, but it's also less expensive ($219.99 vs. $369.99). The WIP330 runs on the
Windows CE 4.2 operating system which enables it to run the CE-version of Internet Explorer for access to just about any web page content. Although, I wasn't able to view my
Linksys webcam which uses an ActiveX control from this mini-IE version.
Linksys claims that you can receive live video on the WIP330 Wireless-G IP Phone when you access any web camera, including the
Linksys WVC54GC Wireless-G Internet Video Camera. I actually just got my hands on a cool
Linksys Wireless-G PTZ Internet Camera with Audio WVC200
that I am reviewing but wasn't able to get it to work with the WIP330. When trying to connect to my home webcam I receive an authentication dialog box and the number keypad won't let me enter in my username/password - it just ignores my key presses as the cursor flashes slowly as though mocking my attempts to access my home webcam. I'll have to contact
Linksys tech support on that one.
InstallationIn any event, installing and setting up the WIP330 was a breeze and in fact I didn't need to read the manual to do it - always a good thing.
Linksys is known for easy, turnkey networking solutions, so considering the complexity of WiFi combined with configuring VoIP SIP settings, I was impressed with how easy the phone was to install. One caveat -
Linksys pre-configured the SIP settings for me, but I checked them out and could have just as easily entered the info myself.
WiFi SettingsAfter first turning on the phone, I simply added my WiFi router's SSID and the WEP encryption key and the WIP330 acquired an IP address, DNS settings, etc. from the DHCP server. Next, I made my first test call by dialing the phone number that
Linksys pre-configured on the device. I dialed 1-415-762-XXXX and the WIP330 rang and displayed the CallerID of the caller. It doesn't support CallerID with Name, however if you add the phone number to the Phonebook and associate a contact record, it will display the name of the contact. The handset stores the last 20 call history records and can save 250 phone book entries. When entering contacts using the 10-digits on the keypad, you have to have fast fingers when entering in the alphanumeric characters or risk it moving onto next letter. If I hesitated for a split second, it moved to the next character. The delay should be user-configurable.
The sound quality was excellent, which is attributable to the G.168 echo cancellation, jitter buffer control, and packet loss concealment. The comfort noise generation (CNG) also helps the voice quality perception. The phone supports the usual codec suspects, including G.729ab, G.711u, and G.711a. While you are on a call, you can mute the call, put the caller on cold and even answer a second call on Line-2. In fact, I initiated a 2nd call to 1-415-762-XXXX from another phone and I heard a series on tones on the ear-piece indicating a 2nd inbound call. I was able to take that call which automatically put the first caller on hold. Simply by pressing the softkey button underneath "Switch" on the LCD caused it to toggle between the two calls. Next, I pressed the softkey button underneath "Options" and was able to select "Conference" which then conferenced the three phone devices together (2 landlines & 1 WIP330 IP connection). In addition, there are two types of transfer features available. You can use the Transfer feature when you want to introduce the caller you are transferring or you can use the Blind TFR (Transfer) feature when you want to transfer the caller without introduction.
Profile of the phone:
ProfilesUp to 10 hotspot/WiFi profiles can be stored and you can designate the order of the profiles saved by the IP Phone. The next time it is powered on, it will use the first profile to automatically connect to the profile's wireless network. If that network is not available, the IP Phone will try the next profile. This will continue until it has connected to a wireless network. It can connect to "open" WiFi networks or secured WiFi networks - both WEP (64/128) and WPA-PSK encryption are supported.
Phone CustomizationYou can customize various settings of the IP Phone including the Ring Option, Keypad Tone, Wallpaper, Date & Time, Language, and Phone Password. You can also upgrade the firmware of the phone over an IP connection. There are five different ring types - ring, ring once, silent, vibrate, and ring & vibrate, as well as (they claim) seven customized ring tones in the documentation, but I only found 3 available. Perhaps they send me an early production model?
Web browserThe web browser is the Windows CE version. Entering in URLs using just a number pad is a bit of a pain, but with a little practice I was able to quickly navigate some sites on the 2.2" LCD screen. Although you can browse the web on the phone, really the browser's primary purpose is to allow you to logon to hotspots, hotel WiFi networks, and other WiFi networks that require some sort of Web-based authentication - often used to ask for billing info. I tested the browser and it worked pretty well, but you'll definitely want to add your commonly visited sites to the Favorites, since I already stated, typing on the number pad is tedious.
I couldn't figure out way to add a currently viewed webpage to Favorites. The toolbar didn't display an icon for adding the current page to Favorites which would make it much easier. Instead, I had to exit out of the browser, go to separate screen and manually type the URL. This is a pain considering how difficult it is to type a long URL along with special characters (: and /) using just a number keypad. Also, if it's a long URL path you may not even remember the exact URL to key in.
When surfing the WIP300, it has an auto-scroll feature so you don't have to repeatedly press the navigation button. You simply press and hold down the center selection key for two seconds until you see an orange navigation cursor. Then you can use the navigation pad to scroll up and down the webpage. To use the cursor as a page up and down button, you simply press and hold down the center selection key for an additional two seconds until the cursor turns blue with a "P" in the center and then use the navigation pad to move up and down the webpage.
Here are some of the menus on the WIP330:
Here's the phone's web admin:
Here are the specifications:• Support SIP v2 Standards
• Compliance with IEEE 802.11b/g Wireless Standard
• Powered by
Microsoft Windows CE, with IE Web browser
• High Resolution Color LCD screen
• Support QoS (Quality of Service) to ensure best quality voice
• Enhanced Power Saving Design for Extended Standby and Talk Time
• 50 hours standby time, 3 hours talk time on average
• 3-Way Conferecing, Call Hold and Resume, and Caller ID
• Fast Hotspot Authentication
• Supports auto-provisioning using HTTP or HTTPS for configuration and upgrades
• Outdoor up to 300m via Embedded Antenna
• One mini-USB Socket, One Stereo Ear Phone Jack
• QVGA TIF 2.2 inch LCD (240*320 pixels) with 65K colors
• 32MB Flash, 64MB SDRAM
• Multiple Access point Registration Support
• TCP/UDP/IP, IPV4, DNS, SDP, ARP, ICMP, DHCP Client, Static IP
• WEP (64/128), WPA-PSK Encryption
• ToS
• SIP v2
Session Initiation Protocol (RFC3261), SDP (RFC2327)
• SIP Session Keep Alive
• G.711( A-law and μ-law), G.729 A
• In-band, Out-band (RFC2833)
• G.168 Echo Cancellation
• Jitter Buffer Control - (default 180ms, max 900ms)
• Comfort Noise Generation
• Packet Loss Concealment
• Speaker and Microphone Volume Control
• VAD - Voice Activity Detection
• 3-Way Conferencing
• Peer-to-Peer Dialing
• Call Hold and Resume
• Caller ID Presentation
• Caller ID Presentation Restriction
• Dial by Phone Number
• Call Forward
• DTMF Tone Detection
• Consultation Hold and Transfer
• Call Waiting and Retrieve
• Mute
• Speed Dial
• Last Number Redial
• Volume Control
• Ringtones: True Tones
• Phone Book (250 records)
• Call History (20 Records)
• Language (English/Spainish)
• Vibrator (Silent mode)
• Password Security
• Date & Time (NTP time synchronization)
• Internet Web Browser (Microsoft IE)
• AES or SSL Encryption
• Firmware upgrades using HTTP, or HTTPS
• Configuration change using HTTP, or HTTPS
• Embedded Web configuration interface (with password protection)
• Power-up Diagnostic
Pros- thin and lightweight
- Web browser surfing capability
- Phone itself had web admin interface
- support for 10 WiFi profiles and built-in support for T-Mobile, Telefonica, and Hinet hotspots
- SIP v2 support
- Pressing +/- during a call will adjust handset speaker volume
- Can sync/connect to the device over USB
Cons
- No POP3 or SMTP email support (though can always use web browser to send or check email)
- You have to have fast fingers when entering in alphanumeric characters or risk it moving onto next letter
- Pressing +/- while not on a call doesn't adjust ringer volume - have to go into settings
- No speakerphone
- Mystery button on its left side that does nothing. Perhaps for future firmware release?
- Only supports 27-bit WPA key
- No
Skype support
- Pricey @$369.999 - you can buy a Windows Mobile 5.0 phone or PocketPC for this price
ConclusionThis
Linksys WIP330 is great for businesses using any IP-PBX that supports the SIP standard, which is pretty much every IP-PBX these days. If you recall, I had an
"exclusive" on Cisco finally adopting the SIP standard (Cisco Unified Communications) over their proprietary Skinny protocol, thus you can even use the
Linksys WIP330 on a Cisco Call Manager system. Of course the
Linksys WIP330 will work with the ever-popular Asterisk phone system as well. No
Skype support is surprising, but I did notice some
Skype config files on the phone itself, so perhaps this capability is coming. Certainly Windows CE has the ability to run a
Skype client. Even with its built-in browser and its ability to logon to authenticated hotspots, due to its price ($369.99), only executives with money to burn, or gadget freaks that just have to have the latest and greatest gadgets, will find the
Linksys WIP330 attractive enough to purchase.
Gadget freaks who just have to have the most feature-rich WiFi phone available today can get the
Linksys WIP330 WiFi VoIP phone from Amazon as well as
from VoIPSupply.com.
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Interesting IP-PBX stats within
Infonetics's Enterprise Telephony report that shows IP-PBXs continued growth even with a generally slower 1st quarter. They also list the current IP-PBX market leaders (Cisco is listed as #1). I'd be curious the stats for "hosted IP-PBXs" that use SIP trunking with no customer premise IP-PBX equipment. My bet is that market is growing as well.
Overall PBX/KTS revenue down 2%, IP PBX up 1% in 1Q06
Despite a seasonally down quarter for the overall enterprise telephony market, the slow and steady move from circuit switching technology to packet switching technology remains evident, with worldwide TDM system revenue falling 11% and IP PBX revenue inching up 1% between 4Q05 and 1Q06, says Infonetics Research in its latest Enterprise Telephony report.
Combined, worldwide TDM and IP PBX systems revenue dipped 2% to $2.1 billion in 1Q06, but is 15% higher than a year ago. Annual revenue is forecast to grow to $11.4 billion in 2009, driven by strong IP PBX sales worldwide as more organizations move to voice over IP. Between 2005 and 2009, IP PBX revenue is forecast to jump 82% while TDM revenue plunges 88%.
"The overall enterprise telephony market was not immune to the first quarter blues, but the IP PBX category managed to eek out a small quarterly gain" said Matthias Machowinski, directing analyst at Infonetics Research. "We recently talked with 450 companies in North America about their voice infrastructure plans, and the results clearly indicate a steady move to VoIP, which will put this market on a nice steady growth trajectory over the next few years."
1Q06 Highlights
- In EMEA, the top IP PBX system line shipment vendors are Alcatel, Siemens, and Nortel
- The top IP PBX system vendors in North America are Cisco, Avaya, and Nortel in a very close race: Cisco was just in 3rd position the previous quarter, and the difference in 1Q06 market share from 1st to 3rd is less than 2 points
- Cisco leads the IP phone market, with 39% unit market share; the next closest competitors are 3Com and NEC, who are tied for 2nd
- Hybrid PBXs account for 63% of PBX line shipments; by 2009, they will account for 78%, up from 61% in 2005
- 45% of PBX/KTS systems revenue comes from EMEA, 30% from North America, 19% from Asia Pacific, and 7% from CALA
Infonetics' report tracks IP deskphones, IP softphones, TDM PBX/KTS systems, and hybrid and pure IP PBX systems and IP PBX shipments by system size (2-40, 40-100, 101-400, 401-1K, and 1K+ lines) in North America, EMEA, Asia Pacific, CALA, and worldwide. Companies tracked include 3Com, Aastra, Alcatel, Avaya, Cisco, Ericsson, Inter-Tel, Mitel, NEC, Nortel, Polycom, ShoreTel, Siemens, snom, Sphere, Swyx, Toshiba, Vertical, Zultys, and others.
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SiteSpeed just released their new beta version (5.0) of their video/voice over-IP software. The beta release includes significantly enhanced video quality, as well as PSTN Out calling and new Voice only PC to PC calling. They are also supporting Macs in this beta. I actually tested an older version of SiteSpeed which had excellent voice and video quality, so I can't wait to check out the latest version.
SightSpeed's Phone Out, can be used to make low-cost PSTN calls from both PCs or Macs to anywhere in the U.S. for 2 cents per minute. You can also make calls to the United Kingdom or Japan for 3 cents per minute. They use
Kayote Networks for termination. SiteSpeed informed me that SightSpeed’s Phone In is coming soon, so SightSpeed users receive calls to their PC or Mac from within SightSpeed. As part of this release, SIP signaling for PSTN termination support is added as well as enhancements to its SOAP based interface for working with partners.
Now if only I could marry the best of
Gizmo (Asterisk support, slick interface, call recording, XMPP support) with the best of SiteSpeed (excellent video quality) along with the ability to interoperate with
Skype and the ability to stream my videos to others (think
Slingbox
or
Orb Networks) I'd have the perfect softphone!
Alas, that day has not yet come, so we still have a fractured market with many softphones each with their stengths and weaknesses.
Nevertheless, the SiteSpeed client is pretty impressive. Just check out the synopsis of the new features:
PC-to-PC-Voice Only CallsThis new mode is perfect for those times a user doesn't want or need video, these voice calls are always free to other SightSpeed members.
Improved VideoSightSpeed significantly enhanced its video quality through the addition of a new advanced beta codec found within SightSpeed by clicking Settings and selecting Optional Settings. The new video codec is the last option on this list and will be standard when the 5.0 release comes available in July.
Enhanced Contact List Display OptionsThe new contact list manager provides multiple ways to see who is online and offline and provides the choices available for how to contact them. They also announced they will be working with Plaxo for contact management integration.
Finally, is it me, or does it seem like everyone in the VoIP space is now integrating with Plaxo for contact management?
Jajah, a VoIP client uses Plaxo. I'm one of the beta testers for AOL PhoneLine and it too features Plaxo integration. Soon Plaxo will become the online Yellow/White Pages to find anyone in the world with instant click-to-call. Once everyone uses them for easy VoIP click-to-call functionality, the
NSA will coming knocking on Plaxo's door - "We want to know who Tom Keating has been talking to. He's Irish, so we suspect he's part of the IRA terrorist organization."
Kidding aside, I have no problem with the NSA tracking who I call. I have nothing to hide. From what I understand they are only tracking numbers dialed and are not recording the content of the calls. After they find suspicious patterns for an individual they then have to get a court warrant. So NSA, you have my permission to track and find the terrorists any which way you can (legally under the Constitution), then send in the Predator drones to kill them dead. (yes, I know 'kill' and 'dead' in the same sentence is redundant).
ACLU members and libertarian readers feel free to blast away on my views of privacy, your interpretation of the Constitution, and the legality of NSA "wiretapping" (though 'wiretapping' is a misnomer since they are acquiring call records). Flame away!
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SIPphone CEO and chairman Mike Robertson wrote me yesterday to tell me that Gizmo 2.0 is now live, and I should try it.I just have, and I like its attractive User Interface and general ease of use.But what's more interesting is something Mike told me.He views this build as in essence, a SJLabs and CounterPath-softphones [...]
Written by Russell Shaw on May 24th, 2006 with no comments.
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Interesting set of findings from an Interop Labs' test of VoIP gear run on the Interop Las Vegas show's earlier this month.Network Address Translation, or NAT, for short, (as shown on th Cisco site, above) can break VoIP. NAT does this by its nature as a procedure that masks private IP addresses from public view. [...]
Written by Russell Shaw on May 17th, 2006 with no comments.
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Siemens Communications Inc. today announced its HiPath BizIP offering, a new peer-to-peer SIP VoIP phone system that negated the need for a complex enterprise network telephone system. Conceptually, this is similar to
Popular Telephony or Nimcat Networks (
now part of Avaya) , but the real news here is the fact that Siemens, a traditional "big iron PBX company" is now offering a P2P phone system that
doesn't require a centralized PBX - instead the intelligence is located on the "peers", i.e. the phones. Now that's
Avaya and
Siemens that have embraced P2P phone systems. Can the death of the big iron centralized PBX be far behind?
Also,
Siemens Communications also announced announced that it would provide development-friendly, business-focused communication applications across a framework of Service Oriented Architecture (SOA) standards and associated Web services interfaces. The
Siemens Communications SOA initiative, which began in 2004, will ultimately encompass all key enterprise innovations and services – including past and future communication solutions.
The HiPath BizIP feature overview:
-- Helps enables low-cost calling via the Internet and ISDN.
-- Supports typical enterprise telephone services such as three-way conferencing, speed dialing, caller lists, call diversion and call swapping.
-- Includes integrated personal answering machine capabilities, with all phones working together to ensure voice mails are not lost when a phone is busy or out of order.
-- Scales to allow from two to 16 phones to be connected as a single workgroup.
-- Can be used in branch offices.
-- Provides a flexible Web tool for simple administration.
-- Includes back-up functionality between telephone terminals.
-- Has quality of service equal to a conventional telephone system.
"This solution stands out for its low investment costs, helping a small office or home office to set up a robust VoIP system without having to create a complex communication infrastructure," said Mark Straton, senior vice president of Marketing,
Siemens Communications Inc. "Installation and maintenance costs are also lower because the existing LAN infrastructure is reused and peer-to-peer software helps eliminate the need for complex telephone configurations."
Communication with public voice and broadband data networks is handled by the BizIP Access Device, which helps ensure that a service provider's VoIP phone numbers can be used for Internet telephony, ISDN lines or both. The HiPath BizIP solution is pre-packaged to interoperate with low-cost session initiation protocol (SIP) services offered by operators. In addition, analog phones, fax machines and door intercoms can still be connected.
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Today as expected, Cisco announced they have finally embraced SIP (Session Initiation Protocol). The platform for doing so will be Cisco Unified CallManager5.0, the call-processing component of the Cisco Unified Communications system.As the last major IP systems and equipment vendor to accept SIP, it does kind of seem that Cisco was dragged kicking and screaming [...]
Written by Russell Shaw on March 6th, 2006 with no comments.
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Pingtel, a provider of open source, Linux-based enterprise VoIP solutions, and
Voxbone, a provider of international VoIP origination services will announce on Monday the completion of interoperability testing between their respective offerings. As a result of this certification, customers can select Pingtel's SIPxchange
IP-PBX VoIP software solution in combination with Voxbone's call origination services and thereby benefit from cost-effective calls routed to a SIP-based device (IP-phone, IP-PBX, etc.).
When a customer is in need of an international presence the customer interconnects via VoIP to the
closest Voxbone POP. Voxbone then allocates the desired amount of numbers and capacity to the customer. When someone calls to one of these numbers Voxbone forwards the call via VoIP to the customer. Unique to the Voxbone offering is that they only charge a
fixed monthly fee for this service - there are no per-minute fees. Here's a diagram explaining the architecture. The POPs are on the left and the SIP device (such as Pingtel) is on the right. The middle is the network/Internet.
Voxbone leases international VoIP virtual phone numbers and worldwide origination services via VoIP to organizations in North and South America, Europe and Asia/Pacific regions. Using either direct inbound dial (DID) or virtual numbers from Voxbone, customers may receive inexpensive, locally dialed phone calls from 50 countries and 4,000 cities throughout the world.
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Camrivox has announced that it has added support for
Google Talk to its range of Flexor IP telephony adapters and IP phones. The Flexor line of products allow users to register a SIP (Session Initiation Protocol) account and a Google Talk account simultaneously on the same VoIP device. The Flexor line also supports analog FXO/FXS ports, so it can also make/receive landline/PSTN calls. Interesting that you can do SIP, Google Talk, and PSTN in/out all from one ATA (analog telephony adaptor). Indeed, it looks like the Flexor 151 and 201 are both nice converged ATAs -- all from a company I just learned about today. They also carry IP phones, such as the Flexor 500, but I didn't see any photos of the Flexor 500 on their website.
"We believe that Camrivox is the first company to add this capability, and our Flexor range is therefore the first devices that enables Google Talk to be used without a PC," said David Moorhouse, VP of Marketing at Camrivox.
The Google Talk capability is supported by Camrivox's complete range of VoIP products including the Flexor 500 IP phone, and the Flexor 201 and Flexor 151 VoIP telephony adapters.
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The iPod is one of the most recognizable products due to it's smooth exterior, elegant shape, and stylish form-factor. So what you do get when you cross an
Apple iPod with a VoIP gateway? Why the
Polypower VRM0-Sxx series of course!
This VoIP gateway clones the iPod's small size, measuring just 100x80x28mm and clones the iPod's original white/ivory color. It includes 1 WAN + 1 LAN + 1/2 FXS + 1 FXO (option). You can use this product with a SIP-based termination service provider along with your favorite analog/cordless/DECT phone. The
VRM0-Sxx features built-in QoS, T.38, auto-provisioning, and many other common features. Based out of Taiwan, this new SIP VoIP gateway features the following specs:
· SIP Protocol
· Codec support G.711, G.723 ,G.726, G.729A/B
· Features : Call Forwarding, Caller ID, Call Hold, 3-way Conference Call ,Toll Bypass
· QoS and FAX T.38 support
· Auto Provisioning
· Small size: 80 x 100 x 28 mmThey also have a
Skype VoIP adaptor available, pictured here:
Soon they will also have a stylish IP phone available which looks like the following:
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I just wrote about Arcosoft’s VoIP call recording software this week and now just a few days later they launched another interesting product called VONaLink ScreenPop. ScreenPop works with any VoIP phone system based on the SIP standard, such as Vonage or Asterisk, to provide screen pops and even the ability to reject calls. Buh-bye telemarketers!
Typically Caller ID is read by a computer using an analog modem or a proprietary CallerID box. VONaLink uses open standards in the VoIP world to simply extract CallerID info by monitoring the network packets.
More important than the ‘techno-speak’ on how it works, using the Caller ID of the incoming call, ScreenPop searches for the caller in Microsoft Outlook contacts, or launches custom applications to search the web or company database. If the caller is found, the information is popped on the screen.
Unwanted callers can be added to the reject list. Integration with Vonage Click2Call allows outbound calls to be placed by clicking in the call log within ScreenPop. Sweet! Now if only I was still a Vonage customer.
VONaLink ScreenPop runs on Windows XP, 2003, and 2000. Priced at $29 USD
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One of the pains of most current ATA devices is that you lock you into a specific vendor by password-protecting the firmware. So even though the ATA is SIP-based, you can only use it with the VoIP provider you initially chose. If you ever want to switch, you're left with an ATA brick. Even if the ATA isn't locked, unless you are a VoIP geek and now how to configure SIP settings, you might be a bit daunted to try and switch providers and leverage your existing ATA investment.
Fortunately, some ATA manufacturers have seen the light. For example, Zoom has an
interesting online wizard utility that helps you figure out the settings for your SIP provider and it will automatically configure their VoIP Freedom Model 5800 ATA device for you. It simply asks you to pick one of dozens of VoIP service providers and then it asks your Zoom ATA's MAC address, which it then auto-configures. It's too bad
Vonage doesn't lock their ATAs. I have a Cisco ATA-186 formerly provisioned by
Vonage sitting in my garage right now.
In addition, Zoom told TMC something interesting: "Zoom releasing the product is important. We currently sell DSL modems in all Staples, CompUSA, Fry's and Micro Center stores in the U.S. We also have retail penetration in international markets. Zoom has significant retail presence and literally decades of experience at retail. Vonage has muddled the VoIP hardware market in the U.S. and UK retail markets, with
heavy slotting fees creating a very artificial model. That will continue in the UK and US as long as Vonage follows their current course. In the markets where the slotting fees are not available, we think VoIP Freedom will be the ascendant model."
In any event, check out the news for yourself which hits the wires tomorrow.
Zoom Introduces VoIP Freedom
Low-cost product lets users choose from dozens of VoIP services
Zoom Technologies, Inc. (NASDAQ: ZOOM) has begun shipping the Model 5800 Zoom VoIP Freedom, a revolutionary product that lets a user choose from dozens of VoIP service providers with an easy Web-based Chooser utility. Most VoIP services lock a user into their service with inflexible hardware. Zoom VoIP Freedom maximizes customer choice by letting the customer review and choose the service best suited to his or her needs, and then automatically configure the Zoom VoIP Freedom device to work with the selected service.
The Chooser includes VoIP services from the Americas, Europe, and Asia, with Web-based menus in languages that include English, Spanish, German, and Vietnamese. Once a customer finds an attractive service, selecting that service automatically configures the Zoom VoIP Freedom device. Changing the service is just as easy, so the customer maintains freedom of choice.
Zoom VoIP Freedom lets customers use their favorite phones. A customer can plug a normal phone into the VoIP Freedom's Analog Telephone Adapter (ATA), and then make or receive calls using that phone. A customer can even plug in a cordless base station, so that the customer can make and receive Internet phone calls with a cordless phone.
VoIP Freedom's Web-based Chooser currently allows selection from a list of 25 VoIP service providers. Zoom invites service providers supporting the industry-standard SIP protocol to register their service for use with the VoIP Freedom ATA at www.zoom.com/partners/itsp . There is no charge for registration.
Zoom VoIP Freedom customers can decide whether to retain conventional landline telephone service along with their VoIP phone service. If they do, the ATA lets them easily make and receive calls through either their VoIP or landline service. Two different calls can take place simultaneously over the VoIP and landline telephone services, providing a "second phone line" capability. Emergency calls, such as 911 calls in the US, are automatically sent over the landline phone service to help assure proper emergency response to the correct location. In the event of a power loss that disables VoIP calling, the VoIP Freedom ATA automatically connects ATA-connected telephones to the landline phone service.
"Two-thirds of broadband connections worldwide are provided by ADSL", said Frank Manning, President and CEO of Zoom Technologies. "Nearly every ADSL line has landline phone service available at little or no additional cost. Zoom VoIP Freedom makes use of that landline phone service to properly handle emergency and alarm calls and to provide powerful enhanced features. For instance, someone with a mobile "cell" phone can easily escape the high cost of calls to other countries. That person just calls the Zoom VoIP Freedom device, which provides VoIP dialtone and bridges the mobile phone user to VoIP. The VoIP Freedom device can even use caller ID to decide which calls it answers; so it answers calls from the appropriate family or business members, but does not interfere with other incoming calls."
Zoom VoIP Freedom has an estimated retail price of $59. Anyone can explore the wide range of VoIP services already available with Zoom VoIP Freedom by going to www.zoom.com/chooser
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SunTec is an interesting company that has been working with telcos to help these companies understand the importance of a singular customer view in the face of increased competition and disparate legacy billing systems. With triple play, quad play, etc. service providers often have legacy billing systems that don't offer a unified view of the customer. These disparate billing systems make it more difficult to offer discounts for loyalty since they often do not know what customers have already.
SunTec stated, "On the customer side, having one bill or being able to speak to a person that knows the entirety of what they're paying for versus talk to three different departments due to siloed systems is of tremendous value." Certainly, with cable companies and phone companies now stepping onto each other's turf, this added competition will only drives price down and margins, which will force them to look for ways to increase efficiency and improve customer loyalty.
SunTec's
TBMS-T system accomplishes a unified billing platform and integrates well with existing systems, is able to take on services as they're added. SunTec just recently deployed their convergent voice billing solution for Cable One, Inc., a wholly owned subsidiary of The Washington Post Company (NYSE:WPO) and operates 52 cable systems serving 685,000 subscribers in 19 states. The solution enabled billing for Cable One's new voice service and provides flexibility to introduce dynamic promotional plans with free-minutes for specific destinations along with value added services and product bundling options. The solution delivers a unified bill for the entire suite of services provided by Cable One. SunTec's Solution supports dynamic tariff modeling and handles Session Initiation Protocol (SIP)-based services like multi-conferencing, video conferencing, and prepaid voice enabling Cable One to introduce new competitively priced product bundles effortlessly. In addition the solution flawlessly integrates with leading telecom and tax databases that include LERG and Vertex.
"This is a win-win situation for Cable One and SunTec! I am extremely pleased with the outcome of the implementation. SunTec's commitment to quality and timelines at competitive prices is remarkable," said Kishore Reddy, Director of Voice Services, Cable One.
"In every way, SunTec met or exceeded our expectations; be it our deadlines, strict budget parameters or the level of quality we demanded, the team more than delivered on every front," said Stephen A. Fox, Vice President of Digital Services, Cable One. "The team demonstrated competence, integrity, and great sensitivity to Cable One's evolving requirements. I would like to see this not as a client-vendor relationship but as the beginning of an excellent partnership," added Fox.
"By rolling out the voice service, Cable One would be among the top league of service providers in the United States. We are excited about being a part of Cable One's success and their quest to be America's Best Cable Company," said K Nanda Kumar, CEO SunTec.
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I recently discovered
Hotty.com, a website that is offering a new SIP VoIP Softphone,
Megapin Inc. has developed a few SIP and Jabber products that they offer to service providers and wholesalers to OEM, with full "skinning" capabilties. One of their prodis
M2-UA, a SIP softphone application and the other is
M2-IM, which has both SIP and Jabber IM/presence support. The M2-IM product will support
Yahoo! Messenger, MSN Messenger, AIM, ICQ, and
Google Talk. Their softphone products feature standard voice codecs including G.711, GSM, and Speex.
I was actually checking out the Hotty.com website last week, but now their homepage shows a 55 day countdown till the product launches. Where'd the content I saw last week go? Fortunately, I keep a very lengthy browser "cache", so I was able to copy the screenshots from my cache folder.
I still haven't figured out why Megapin, Inc. is using the domain name
hotty.com for a softphone application. Sounds like a porno site if you ask me. And if you look at the black background with the coundown clock it certainly has the feeling of a porn site to me. Either that or a hacker site, which often use a black background. Black is cool in hacker world. Maybe their softphone application will be targetted at online dating sites or perhaps sites with female "
hotties" taking inbound VoIP calls for some Phone Sex over IP (after you give your credit card number of course)
Well, only 55 more days till we find out what this is all about...
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Editor's Note: Glad to see Verizon is embracing a decent protocol. Kudos
Wholesale VoIP customers are now receiving new cost saving options from Partner Solutions. Leveraging its extensive reach with an innovative suite of wholesale VoIP services, Verizon Partner Solutions is offering a new cost-effective option for VoIP providers, using SIP (Session-Initiated Protocol) Gateway Services.
Under this program, Verizon SIP Gateway customers opting for OCN-based metered plans can enjoy a reduced rate for calls from their end-users to select Verizon Business local customer telephone numbers (those utilizing legacy MCI local switches).
SIP is an Internet-based signaling protocol used for creating, modifying and terminating sessions between two or more users on the Internet, including Internet telephone calls, multimedia distribution and multimedia conferences.
“This is simply another way we’re making it easier for our wholesale customers to do business with us - by listening to them and continually evolving our offerings to help them help their customers,” said Tom Maguire, senior vice president, Verizon Partner Solutions.
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Shares in NetComm soared 15% on news it had established a partnership with NEC Business Solutions. The broadband technology specialist said it is set to develop and manufacture the first fully approved third -party SIP-compatible digital telephone to operate with NEC’s UNIVERGE SV7000 IP Telephony solutions.
MD David Stewart said that NEC Corporation, a world market leader in telephony, were looking for a business-grade telephone that could support the many functions of its UNIVERGE IP-PBX range. “TheNetComm Product Customisation Division was commissioned to build the V95 for NEC to deliver a standardised product on the SIP platform and to capitalise on the powerful feat ures in the UNIVERGE SV7000 IP telephony solution,” he explained.
The NetComm V95 IP Telephone uses the popular Voice over IP standard Session Initiation Protocol (SIP) and will be now be used by NEC. Evaluation samples of the V95 have also been sent to several other countries where NEC operates, to review the phone’s compatibility for overseas use, the company advised. NetComm said that they are extremely optimistic of broad adoption worldwide.
Source: egoli
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Starent Networks, Corp., a global market leader in mobile packet core solutions, today announced its Session Control Manager – an integrated Session Initiation Protocol (SIP) Proxy/Registrar, Proxy Call Session Control Function (P-CSCF), and Policy Agent – for its ST16 Intelligent Mobile Gateway. Starent Networks’ Session Control Manager (SCM) provides IP Multimedia Subsystem (IMS)/Multimedia Domain (MMD) capabilities while providing a migration path to a complete IMS/MMD architecture. The Session Control Manager is a software upgrade to the company’s industry leading ST16 Intelligent Mobile Gateway. The standards-based solution simultaneously supports an IETF SIP Proxy/Registrar, a 3GPP/3GPP2-compliant P-CSCF, and a Policy Agent.
“With this solution, we are evolving the mobile core network towards the IMS/MMD architecture,” said Ashraf Dahod, CEO of Starent Networks. “As a result, mobile operators will enhance their subscribers’ experience and be able to deliver voice, data, and multimedia over IP.”
The introduction of the Session Control Manager represents the first multi-access solution with integrated SIP/IMS features, high performance, and unparalleled reliability. By integrating the Session Control Manager into an intelligent mobile gateway, the system acts as the critical first hop for SIP/IMS applications, enabling end-to-end security, policy control, enhanced charging, and mobility.
“Integrating key IMS functions into existing networks elements is a logical approach to deploying services now while preparing mobile networks for a migration to a full IMS architecture," said Robert W. Johnson, Sr. Analyst, Telcom/Datacom with Venture Development Corporation (VDC). “With significant deployments in the mobile packet core network, Starent Networks is well positioned to deliver appropriate SIP and IMS functions to the mobile wireless market.”
In order to showcase its Session Control Manager, Starent participated in the Global MultiService Forum’s global multi-vendor interoperability IP Multimedia System (IMS) trials, also known as GMI 2006. During the two-week event, Starent’s Session Control Manager successfully participated in testing key interfaces, protocols, and procedures defined in MSF Release 3 Architecture, as well as 3GPP/3GPP2 IMS/MMD.
The Session Control Manager simultaneously supports IETF SIP software clients and IMS/MMD software clients, allowing operators to add the solution as a standard SIP Proxy. An operator can deploy a mix of SIP-based and IMS-based applications, such as Voice over IP (VoIP), Push to Talk over Cellular (PoC), mobile TV, Instant Messaging Presence Service (IMPS), and Voice Call Continuity applications while evolving their IMS core.
The Session Control Manager is combined with Starent’s market-leading, multi-access intelligent mobile gateway, allowing operators to deploy access-independent connectivity and global roaming to a multitude of technologies, such as CDMA 1X/1xEVDO, UMTS/W-CDMA, WiFi, WiMAX, and others.
Source: Starent
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IVR Technologies, a software developer for IP enhanced services and real-time billing solutions, today announced the availability of Talking SIP 3.1. Talking SIP 3.1 represents the culmination of a tremendous investment by IVR Technologies in the development of its third generation architecture to further entrench IVR Technologies' leadership position of providing industry leading and revenue-generating applications with the utmost in real-time billing control and flexibility, without sacrificing ease-of-use nor low administrative overhead.
"Talking SIP's signaling, media processing and application scripting components have been completely redesigned to reach new milestones in scalability, performance and control for the next-generation network. We are extremely excited about this release as it provides an amazing foundation upon which to build voice 2.0 applications that highlight the empowerment, flexibility and control that broadband access and voice over IP have to offer", said Randall O. Walrond, Product Development Manager, IVR Technologies, Inc.
Talking SIP 3.1 is built around a completely redesigned application, media and signaling architecture to serve as the foundation for rich, in-demand applications that capitalize on the ubiquity, access and richness of broadband connectivity.
The following is a brief overview of the features and functionality embodied within this release:
o International and Web Callback - This optional application allows providers to drive revenue by allowing their customers the ability to trigger callback calls to get a remote dial-tone via SMS, email, telephone, web page or integrated into third party applications via an API to keep customer traffic within your network even when they are traveling out of city, state/province or country. With many creative billing options, this application can even navigate through PBXs and ACDs for use from hotels, motels and corporate offices.
o Reminder/Wake-up Service - This optional application allows providers to offer wake up and reminder services to their customers for the small office/home office as well as to the hotel, motel and timeshare markets.
o Improved Performance - Talking SIP 3.1 offers improved performance, reduced CPU load and a reduced memory footprint to allow the application to make better use of the host system's resources and to allow greater levels of responsiveness and scalability. In addition, Talking SIP 3.1 provides greater access to signaling and media to allow greater control and performance tuning for different hardware platforms in addition to voice caching to help optimize text-to-speech playback.
o Improved Tones - Talking SIP 3.1 adds the ability to play customized ring tones in two-stage calling applications like the Debit Module. This new features allows better internationalization by allowing providers to configure country specific tones like ringing, busy and fast busy to provide their customers with additional levels of comfort when using the system. These tones can be customized to allow opportunities for branding as well as personalization.
o Asynchronous Voice Prompts - Talking SIP 3.1 now supports asynchronous voice prompts to help streamline call flow and reduce hang times. Now you can be playing voice prompts to the caller while background operations like a credit card or voucher recharge are taking place.
o Multi-leg Billing for Callback - Talking SIP 3.1 offers the industry's most accurate multi-leg billing engine with dynamic balance updates and on the fly maximum call durations to ensure that callers receive the maximum call time while enforcing all of the credit limits of their accounts.
o Supports any Codec - Talking SIP 3.1 supports any voice or video codec between calling and called parties to allow proprietary and/or bandwidth appropriate voice and video codecs to be used as well as dynamically changed during a call.
o Open XML interface Added for Credit Card Processing - Our optional Credit Card Verification Server module now supports Verifone's PC Charge Software, LinkPoint as well as an XML interface to allow third party integration to any banking or financial institution. This module allows for automated web based subscription, as well as web, automated and telephone based recharge.
Support and Availability
Talking SIP 3.1 is available for download from the IVR Help Portal for all customers who are within a valid support agreement.
Source: IVR Technologies
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Paradial today announced the general availability of Paradial's RealTunnel 2.0. In addition to making it simple for customers to deliver guaranteed and secure connections across networks, geographical areas and organizational boundaries, the new version enhances the award winning product with new features and standards-based provisioning capabilities on a significantly reduced footprint.
"RealTunnel 2.0 continues to deliver the best possible voice and video quality by automatic and intelligent discovery and use of optimal transport mechanisms for the call at hand. Using the RealTunnel SDK allows customers and partners to focus on delivering communication and collaboration solutions without compromising security or having to solve the complex challenges of connecting calls over firewalls and NATs", said Espen Skjæran, CTO, Paradial.
New features and capabilities include:
Support for ICE/STUN/TURN, UDP, TCP and HTTPS relay support including HTTP proxy and authentication schemes. Small client footprint. Available as SDK for softphones and hardphones, and as a stand-alone enterprise proxy for corporate networks. SOAP provisioning interface. "The new RealTunnel release demonstrates Paradial's continued investment and commitment in making it simple and secure to build, deploy and operate standards-based IP-communication and collaboration solutions", said Ingvar Aaberg, CEO, Paradial.
Availability and Platforms
RealTunnel SDK is available on Windows and Linux. Support for additional platforms is planned.
Key Product Facts:
• The only truly open solution on the market supporting any SIP client and any SIP Registrar.
• Small footprint SDK.
• Most comprehensive FW/NAT product available. RealTunnel supports voice, video and T.120 application sharing across any firewall.
• The customers can use existing network infrastructure firewalls.
• No network or firewall modification is required.
• Excellent voice and video quality.
Supported network protocols:
• UDP
• TCP
• HTTPS
• RTP
• RTCP
Supported standards:
• SIP (RFC3261)
• STUN (RFC3489)
• TURN
• ICE
• Symmetric Response (RFC3581)
• Extension Header Field for Registering Non-Adjacent Contacts (RFC 3327)
• Locating SIP Server (SIP DNS)
The most common HTTP proxy authentication schemes are supported:
• Basic authentication
• Digest authentication
• NTLM authentication
• Proxy pac scripts
Source: WebIT PR
Written by Dal on January 1st, 1970 with no comments.
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Ingate Systems, which develops firewall technology and products that enable SIP-based live communication for the enterprise while maintaining control and security at the network edge, announces that
Sterling National Bank, the principal banking subsidiary of financial holding company Sterling Bancorp, has chosen the award-winning Ingate SIParator to enable SIP trunking.
Sterling, which pioneers the use of new technologies to improve business processes and efficiency, is using SIParators at its Wall Street operations center and Woodbury, NY office to protect its IP-PBXs and to connect to a SIP Trunking service provider, resulting in a two-thirds reduction of its telecommunications expenses.
SIP trunking rapidly reduces telephony costs by using the Internet instead of the PSTN to carry communications traffic (voice, etc.) as far as possible, in effect eliminating long-distance telephony charges. SIP trunking also enables an enterprise to use just one connection for both voice and data, eliminating the cost of redundant network connections.
The SIP trunking solution bridges between VoIP and the PSTN. Sterling, like other enterprises, can now use VoIP to call people who still use traditional telephony service. With SIP trunking long-distance calls are routed over the Internet and terminated at the service provider's PSTN gateway which is located closest to the person being called. The result is a savings for Sterling of approximately $1,000 per month, or 2/3 of their previous telephony costs.
In addition, using SIP trunks eliminated the need for Sterling to purchase local PSTN gateways and costly ISDN BRIs (Basic Rate Interfaces). The company still uses PRI (Primary Rate Interface)-rate ISDN for incoming calls, 911 and toll-free calling but the expense is modest since all outgoing calls that are billable are sent to the SIP trunking service provider.
With such a rapid reduction in communications expenses, Sterling expects that the SIP trunking installation, including the IP-PBX, the Ingate SIParator and the trunking service will pay for itself in less than six months.
"As a company committed to a high-tech, high-touch experience for our customers, SIP trunking helps us maximize the opportunities available with IP communications," said Eliot S. Robinson, Executive Vice President, Sterling National Bank. "Ingate's SIParators were the ideal solution to allow us to deploy SIP trunks. They were fully interoperable with our IP-PBX, the SIP trunking service provider and our existing security infrastructure and perhaps most importantly they just worked."
Ingate SIParators work in parallel with a company's existing firewall to enable businesses to utilize SIP-based VoIP. They solve the Network Address Translation (NAT) traversal issues that are faced by businesses using SIP. They also provide the advanced routing capabilities necessary for enterprises to easily connect to carriers' SIP trunks. And for remote users, small branches and mobile workers, Ingate's Remote SIP Connectivity capabilities allow these workers to use the centralized IP-PBX and the SIP Trunking service to make and receive calls.
"Sterling Bank's experience highlights the benefits of proven SIP applications that streamline business processes and improve the bottom line," said Steven Johnson, President, Ingate Systems. "We are proud that Sterling chose Ingate to enable their SIP trunking deployment."
Sterling plans to expand its use of the Ingate SIParators and SIP trunking to other offices throughout the Sterling Bancorp network. They also plan to extend their use of IP communications beyond voice traffic, to include IM, presence and other productivity-enhancing applications.
Source: InGate
Written by Dal on January 1st, 1970 with no comments.
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Full Disclosure: We use this service in our office via IAX termination. So far it has been pretty darn good.
Building on the momentum of its PSTN Gateway offering, Junction Networks today officially announced the release of its Hosted PBX service and SIP For Business services. These new services break the mold of per-seat pricing by offering an unlimited number of users for Hosted PBX services. Additionally, the SIP For Business service allows free SIP calls with business-class PBX features.
Now, businesses of all sizes can get free SIP calling along with free extension dialing and free customized SIP addresses. For additional fees, customers can add services such as voice mailboxes, pstn calling and auto-attendants. The platform is entirely SIP Address based so customers are free to use any SIP client IP phones including SNOM, Polycom, Eyebeam and Grandstream phones.
"With no per user fees for the service, this is a first of its kind" said Michael Oeth, CEO. "A company with 10 users can now get a fully functional virtual PBX with voicemail, auto-attendant and hunt groups for about $50 a month plus minutes. The competition is still charging north of $50 per user for 'unlimited' minutes that often go unused."
The company is focused on value added features beyond the standard PBX set. For example, customers can add a popular CRM feature called Click-to-Call which allows customers to add a button to their website and give visitors the ability to receive an immediate phone call from the Junction Networks customer. The service then notifies the Junction Networks customer that a call has been requested and the website visitor and Junction Networks customer are immediately connected. Click-to-Call is free to all Hosted PBX customers.
"We have worked hard to build on completely open standards including SIP" stated John Riordan, CTO. "As a result, we can add features such as a Click-to-Call at a fraction of the price charged by competitors. Additionally, our standards-based API allows anyone to fully integrate their offering into our product."
As always, Junction Networks requires no commitments to use the service. "Customers can sign up and use the service for 1 day or 1 year and feel free to close their account if they are not satisfied," stated Robert Wolpov. "We are confident that customers will be amazed at the value they get for our industry low pricing."
The Hosted PBX bundle is free for 30 days. Customers are free to try the service and cancel at any time with no penalty. The service can be used with any SIP compatible IP phone or software phone.
Written by Dal on January 1st, 1970 with no comments.
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Nuance Communications has embraced the standard signaling system for IP telephony to deliver a less expensive system for dialing by voice. The SpeechAttendant Internal Dialer, like earlier products for circuit-switched and IP phone systems, lets employees call their coworkers by just tapping one number and saying the name of the person they want to talk to.
The new product supports SIP, which is rapidly being adopted by IP telephony vendors, making it easier for Nuance to port its technology to each vendor's platform, according to Pat Delgesso, director of Auto Attendant Solutions at Nuance.
The dialer is already compatible with infrastructure from Cisco, Avaya, Nortel Networks, and other vendors.
SIP is helping to open up the enterprise telephony market, once dominated by proprietary products, to a variety of vendors that can introduce new capabilities, said IDC analyst Abner Germanow.
Faster, easier development means lower cost, Delgesso said. At less than $20,000 for a four-port system that recognizes 1,000 names, the new product costs about one-third less than its existing product for IP phone systems that use TAPI (Telephony Application Programming Interface), he said.
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Jabber, Inc. today announced the general availability of the 5.2 release of the Jabber Extensible Communications Platform (Jabber XCP).
The highly programmable, scalable, and secure presence platform enhanced its industry leadership position with the addition of:
- A Session Initiation Protocol (SIP)/SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) Gateway delivered on a new code base providing for transparent federation with SIP/SIMPLE-based messaging systems such as IBM Lotus Sametime, Microsoft Live Communications Server, and the AOL Instant Messenger service. A cornerstone of the company’s multi-protocol approach to real-time messaging, the new SIP/SIMPLE implementation is now available on the Microsoft Windows Server platform in addition to Solaris and Linux.
- Sun Solaris 10 support, the platform on which recent loads tests (http://www.jabber.com/index.cgi?CONTENT_ID=1080) confirmed Jabber XCP’s ability to scale past a million concurrent users. Jabber XCP is also available for the Solaris 9, Red Hat Enterprise Linux 3.0 and 4.0, Microsoft Windows Server 2000 and 2003 platforms.
- Increased security integrity of Jabber XCP through obscured password storage in encrypted files.
- Stanza optimization, through the implementation of XMPP Extension Protocol 0033 (XEP-0033), which provides a method for both clients and servers to send a single stanza and have it delivered to multiple recipients. The benefit is reduced network traffic in situations such as multi-user chat and load balancing among clustered Jabber XCP routers.
- Auto-include Special Interest Groups (SIGs)/Schemas, which simplifies the job of developers by enabling new code to run on Jabber XCP without editing configuration files.
“Jabber XCP continues to lead with the world’s most scalable, programmable, and interoperable messaging and presence platform,” said Dave Uhlir, vice president of marketing and product management at Jabber, Inc. “Today’s release is a significant milestone towards delivering a truly multi-protocol platform that can surpass a million concurrent SIP, XMPP, and/or IMPS users on modest hardware infrastructure.”
"Most people are aware of presence information because of instant messaging. However, over the next several years, presence information will be used to identify when different types of resources including people, conference rooms, vehicles, deliveries, inventories, etc. are available, now or at an estimated time in the future," said Mark Levitt, vice president for collaborative computing and the enterprise workplace at IDC. "In preparing for this future presence-aware real-time work environment, organizations must look for a presence engine that has enterprise class reliability, scalability, security, and extensibility."
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TECORE Wireless Systems, announced the launch of its pre-IMS VoIP/SIP enabled GSM base station to be showcased at this year’s 3GSM World Congress. The BTS-4000RM delivers up to 64 high powered GSM/GPRS/EDGE voice and data channels occupying as little as 5U of a standard 19” telecommunications rack. The BTS-4000RM base station eliminates the need for traditional BSC, TRAU and PCU elements, embedding these functions within the base station platform.
The BTS-4000RM may be integrated directly with TECORE’s SoftMSC as well as other mobile core network solutions over IP, TDM, or IP over TDM. Moreover, the BTS-4000RM is enabled to support TECORE’s exclusive
AirSite Backhaul Free base stations, allowing operators to deploy ‘plug and play’ coverage extensions without the need for wired backhaul links. Due to its small size and flexible interface options, the BTS-4000RM is also ideally suited for remote towns and villages that may require satellite backhaul as well as other applications where macro RF coverage and performance is needed.
With the launch of the BTS-4000RM, TECORE also introduces network interface enhancements to its GSM base station product line. This includes the internal translation from a traditional GSM RAN to an all IP SIP-enabled network interface.
Thus, the BTS-4000RM supports direct connection to a pre-IMS network. Added capabilities include transcoder-free operation to minimize network bandwidth, and direct routing of local IP call traffic to optimize overall backhaul utilization. Specifically targeting applications requiring compact radio access solutions, the BTS-4000RM offers flexible network integration and deployment capabilities while providing macro RF coverage, capacity and performance.
“The operators to whom we’ve delivered and demonstrated the BTS-4000RM are very excited about the flexibility this solution offers,” said Terry Williams, Chief Technology Officer for base station products. “Large capacity, flexible interfaces, and more functionality in a smaller, easy to deploy platform are key to many specialized mission critical applications, and this platform delivers.”
“We are very pleased with customer response to this new product development and technology innovation,” said Doss McComas, Vice President, Business Development at TECORE. “This product builds upon our company’s ongoing commitment to advanced, smaller, more powerful base station solutions for the market.”
The BTS-4000RM will be on display at the 3GSM World Congress 2007 in Barcelona, Spain, February 12-15, 2007, at the TECORE Wireless Systems pavilion located in Hall 8, stand 8C78.
Written by Dal on January 1st, 1970 with no comments.
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