VoIP Monitor - SIP

The SIP Forum Announces Dates and Opens Registration for Third Annual SIPNOC

VoIP Monitor — Mon, 22 Oct 2012 21:05:54 GMT

The SIP Forum announces it will host the third annual SIP Network Operators Conference (SIPNOC 2013), an educational conference focused on the challenges and opportunities of deploying SIP-based carrier services worldwide. Building on the success of past conferences, this year the SIP Forum will add an extra day of special workshops and sessions, designed to educate service provider technical staff on best practices and strategies for the successful implementation of SIP-based services and applications.

SIPNOC 2013 will be held at the Hyatt Dulles Hotel in Herndon, Va., from April 22-25, 2013, and will continue its practice of providing leading technical and operations personnel from the global carrier community with educational, non-commercial and technical content focused on the real-world challenges of deploying SIP services in global IP networks.

Early bird registration is available at http://www.regonline.com/sipnoc2013. More details about SIPNOC 2013 are available on its conference website at http://www.sipnoc.org.

"The growth and acceleration of SIP services within the international service provider community - and the accompanying technological, logistical and businesses challenges - was the original impetus behind SIPNOC and remains our guiding motivation," SIP Forum Chairman Richard Shockey said. "We're pleased to bring this unique gathering for the network operator community back for a third year, once again providing a non-commercial, technically-oriented setting for discussing and sharing ideas and knowledge about SIP implementation."

SIPNOC 2013 will once again bring together communications industry leaders to learn, discuss and formulate new ideas and strategies to address the challenges and opportunities for SIP-based carrier services in fixed and mobile IP network environments. The conference, which is designed specifically for SIP network operations personnel and engineering staff, will feature well-known industry speakers and a number of highly technical educational and instructional panels and sessions. The SIP Forum will also host networking events at the conference and offer a series of informal "Birds of a Feather" sessions, which encourage discussion on a variety of topics in hallways, available meeting rooms and break-out areas.

"SIPNOC demonstrates the SIP Forum's ongoing commitment to serving as the industry's non-commercial think tank for international SIP interoperability and deployment," said Marc Robins, President and Managing Director of the SIP Forum. "Our third annual conference will expand upon key issues identified in previous discussions and through our SIP Forum Task Groups, with the aim of delivering high-level technical support and guidance to the broad service provider community worldwide."

SIPNOC 2013 will build on critical themes discussed at past events, including: application development and testing; SIP trunking interoperability and the SIP Forum's SIPconnect 1.1 technical specification; Fax over Internet Protocol (FoIP) interworking; implementing SIP over IPv6; user-agent configuration; emergency services; policy servers; security; operational issues; call routing and peering; troubleshooting and monitoring; SIP Interconnection; HD-Voice Deployment Challenges; and Video interop issues.

"SIPNOC is an international gathering where communications engineers and network professionals can discuss and troubleshoot the real-world intricacies of working with SIP every day," Robins added. "We expect this year's conference to build on our past successes, and explore new complexities as we strive together, as a community, to deploy SIP-based services in diverse network environments worldwide."

Attendees at SIPNOC 2013 will include telecommunications providers, major backbone operators, interconnect and wholesale solution providers, ISPs, ITSPs (Internet Telephony Service Providers), cable operators and wireless network operators, as well as large enterprises deploying major SIP initiatives. Industry stakeholders - such as network equipment vendors, government agency representatives and academic research organizations - are also encouraged to attend.

The SIP Forum enjoys an international reputation for developing key educational events around SIP deployment. The organization's SIPit series of interoperability testing events regularly provides a test bed for SIP-based applications and equipment and has been heralded as critical for the development of new products and services in the industry. The SIP Forum is also home to committees and task groups comprised of industry experts examining a myriad of SIP-related topics, including the use of SIP with FoIP, video, SIP over IPv6 and user-agent configuration. SIPNOC 2013 Corporate Sponsorship Opportunities

Broadvox Meets Increased Demand for SIP Trunking

VoIP Monitor — Thu, 18 Oct 2012 20:46:43 GMT

Broadvox announces an extension of an aggressive new promotion for its commercial SIP Trunking service. Broadvox guarantees a free 5-day installation period to turn up service on its industry leading SIP Trunks plus a 20% discount off the price of each line.

SIP Trunking saves businesses up to 70% over the cost of traditional PRIs or analog phone lines. By combining voice and data over a single broadband circuit, SIP Trunking allows customers to use broadband more efficiently and guarantees that customers buy just what they need rather than being forced to purchase a 23-channel PRI. SIP Trunking is extremely scalable so companies can purchase lines as they need them. As the alternative to traditional telephony, business SIP Trunking is projected to grow by 15% per year, as more businesses choose to take advantage of the savings and benefits of Voice over IP via an IP PBX.

Through is robust private national network, Brodvox has been offering Voice Over IP services since 2001. As a veteran SIP Trunk provider, Broadvox has established interoperability with the majority of PBXs on the market, making it easier for customers to make the switch from their current phone system, and start saving money right away. Also, by using an Integrated Access Device, virtually any TDM PBX or legacy CPE phone system can be updated with SIP Trunking. Additionally, Broadvox SIP Trunking has a failover and redundancy process to ensure reliability.

“Broadvox continues to meet the demand of the marketplace for VoIP. We have had a tremendous response to our current SIP Trunking service and recent promotion,” said Bruce Chatterley, President and CEO of Broadvox. “The free 5-day installation guarantee is a promise to our customers that we understand their need for a speedy transition from their current phone solution, with minimal disruption to their business.”

Edgewater Networks Certifies Panasonic for its Plug & Dial Alliance Program

VoIP Monitor — Thu, 18 Oct 2012 20:45:49 GMT

Edgewater Networks announces that Panasonic’s SIP communications-based phones have been certified as interoperable with Edgewater’s Plug & Dial solution. The Plug & Dial solution uses the EdgeMarc ESBC and the EdgeView VoIP Support System to automate the provisioning of a wide variety of IP phones. This automation reduces operating expenses and improves the end-user experience for service providers delivering cloud communications services.

Panasonic phones that have been Plug & Dial certified on EdgeView version 11.7.2 include KX-UT113B, KX-UT123B, KX-UT133B, and KX-UT136B.

The Plug & Dial Alliance program provides interoperability testing for multi-vendor VoIP networking environments and automated setup of many leading brands of SIP-based IP phones. Service providers use Edgewater Networks’ Plug & Dial solution to significantly shorten hosted PBX installation times and simplify ongoing moves, adds and changes. The solution uses intuitive voice prompts provided to the end-user so they can “self provision,” completely eliminating pre-staging or manual configuration of IP phones. The solution also provides notification to existing OSS or billing systems at the completion of the automated IP phone configuration. The level of automation provided by the Plug & Dial solution reduces installation times from hours to minutes.

The EdgeView VoIP Support System is used for the ongoing maintenance and management of IP phones. Qualifying phones report call quality scores to EdgeView where they are combined with results from other EdgeMarc monitoring points in a VoIP network. This greatly reduces problem-resolution times and enables service providers to deliver an improved customer experience. EdgeView is also used to remotely administer IP Phones and includes features such as the modification and backup of IP phone configuration files.

The EdgeMarc ESBC and EdgeView VoIP Support System are a part of a comprehensive solution from Edgewater Networks that connect, protect, optimize and manage IP-based communications.

Edgewater Networks and Panasonic will be exhibiting at BroadSoft Connections October 21 – 24, 2012, at the Westin Kierland Resort and Spa in Scottsdale, Ariz. Visit Edgewater Networks at booth #16 and Panasonic at booth #13 and 17.

SIP Certifications, an Industry Standard at 1000+

VoIP Monitor — Tue, 16 Oct 2012 21:11:14 GMT

The SIP SchoolT has awarded the 1000th SIP School Certified Associate. The SSCA is the leading vendor-neutral certification for networking professionals that is globally recognized and has become the SIP certification to attain.

Sreekumar G, the 1000th recipient says, "I'm currently working as a Technical Manager in AVAYA. My primary job includes supporting the AVAYA customers around the globe who are using Unified Communications solutions. The SSCA SIP certification was very highly structured and it's helping engineers in understanding the SIP concepts, call flow, etc. I would strongly recommend engineers who are working in the SIP domain to try this as it will really help in your day to day work."

SIP certification involves a series of online SIP training modules followed by an extensive online test of the technologists understanding and application of SIP. Becoming a SSCA demonstrates that the recipient can work effectively in the VoIP environment and SIP's migration into support Unified Communications. The SIP training program is specifically designed to not only educate on SIP but teach people how to work with confidence in situations that may be completely new to them. Use the discount code "1000SSCA" when purchasing the training and taking the test.

Sharon Golan, Senior Technical Program Manager at Avaya believes that "The SIP training program from The SIP SchoolT has proven to be a great resource for us in getting our engineers up to speed on the protocol that we are building into the Avaya products. Getting engineers certified as an SSCA has brought numerous benefits for us as an employer, not only getting our engineers the required skills, but we can also measure peoples development and reward them as such. For the engineers themselves, they are happy to achieve certified status and be seen by their peers and clients as an expert on SIP. Of course, our customers are benefiting as support queries are dealt with quicker, and they continue to get the excellent support that they are used to. "

Certification can help the employee in many ways including job prospects, job promotion, higher standing in the telecom community along with the confidence that their training and certification gives them the skills to adapt as SIP evolves. When hiring, 68% of IT managers regard certifications as medium to high priority according to the largest vendor-neutral certifying organization, CompTIA.

Getting certified is fast and easy. Networking professionals can obtain the SIP training and then go on to take the SSCA test. If the professional already has the knowledge, the SSCA test can be taken separately without taking the SIP training modules. The SIP SchoolT course descriptions can be found at http://www.thesipschool.com/courses/view which has course outlines and demos. Advice for selecting the best SIP training for the professional can be found at http://www.thesipschool.com/whichtypeareyou.html.

The SIP SchoolT is the web's leading SIP training and Certification service that has quickly gained the backing of manufacturers and organisations that are prominent in the Telecoms industry. SIP, which stands for Session Initiation Protocol has become the most important protocol in Voice and Unified Communications today as it is built into handsets, PBX systems, gateways, and is also displacing traditional digital and analog lines with SIP trunks. Its importance to the Telecoms industry cannot be understated. Educated professionals are crucial to its development, hence the success of The SIP SchoolT programs.

With clients and partners such as Cisco, Avaya, Toshiba, Mitel, Panasonic, Acme Packet, AudioCodes, Bell Canada, British Telecom, NEC and many others it's easy to see why The SIP SchoolT has become the provider of choice for education and with the backing of the TIA, it is firmly established as the world leader.

"The Telecommunications Industry Association, the leader in advocacy, standards development, business development and intelligence for the information and communications technology industry, has officially endorsed The SIP SchoolT as the provider of choice for training and certification for SIP. "

For a full list of training programs plus demos and pricing, please visit www.thesipschool.com.

Sonus Introduces New SIP Trunking For Dummies and Session Management For Dummies Reference Books

VoIP Monitor — Fri, 05 Oct 2012 20:18:55 GMT

Sonus Networks has again partnered with John Wiley & Sons, Inc., the publisher of the For Dummies book series, to produce two new books on how companies can take advantage of the transition to SIP based communications environments. These new books are:

SIP Trunking For Dummies, Sonus Special Edition (Wiley, 978-1-118-48767-9, September 2012)
Session Management For Dummies, Sonus Special Edition (Wiley, 978-1-118-47072-5, September 2012)

Both books are available today and complement the Session Border Controllers For Dummies, Sonus Special Edition (Wiley, 978-1-118-37742-0, May 2012) book released earlier this year.

SIP Trunking For Dummies overviews how SIP trunking, and the move to VoIP, can collectively reduce traditional enterprise telecom bills by up to 75%. A July 2012 report commissioned by Sonus and conducted by Webtorials, found that one in three enterprises have deployed SIP trunking with an average cost savings of approximately 33%.

Session Management For Dummies outlines how a SIP based environment can be leveraged to deploy applications like Unified Communications and video across disparate network environments. Recent Webtorials reports for Sonus found:

SIP-based Unified Communications infrastructure could reclaim 23% of the productivity lost on inefficient communications in large Enterprises.
VoIP (89%), Unified Communications (69%) and video conferencing (65%) are the most important types of media to be controlled via SIP.

Overcoming varied network topologies is a common challenge in many businesses and Session Management For Dummies offers the insight needed to start deploying communications applications in a manageable way across locations and devices.

To download electronic copies of SIP Trunking For Dummies, Session Management For Dummies and Session Border Controllers For Dummies, go to www.sonus.net/dummies.

Vertex Telecom Selects RedShift Networks as the VOIP/SIP Security Solution for Enterprise and Wholesale Customers

VoIP Monitor — Tue, 02 Oct 2012 21:20:15 GMT

Vertex Telecom has selected RedShift Networks to provide their Unified Communications, Collaborations, VoIP and Video UCTM Security technology; which is currently being switched on Vertex Telecom VOIP/SIP network. Additionally, Vertex Telecom has selected RedShift Networks to develop three custom program models for wholesale into three sectors: Financial, Healthcare and Government. The three additional models will work in conjunction with the “core” and “edge” products installed at Vertex Telecom.

RedShift Networks has deployed its UCTM appliances as a first step to further protect and lock down the “core” and “edge” of Vertex networks. According to Amitava Mukherjee, president, CEO and founder of Redshift Networks, “Our UCTM appliance security solution includes a patented behavioral learning engine that automatically gathers network intelligence identifying potentially dangerous vulnerabilities. Coupled with Redshift’s global threat signature database and blacklisting service, Vertex now has a comprehensive UC/VOIP threat management solution built into its core network.”

“We are very excited with our security solutions being implemented into award winning facilities like Vertex Telecom’s. We expect to grow our enterprise business with Vertex into the Philippines and Taiwan telecom markets with the rest of this year and in 2013.” RedShift Networks, Robert Barker, Regional Sales Director, RedShift Networks.

ShoreTel Validates AudioCodes’ Enterprise Session Border Controller for SIP Trunking

VoIP Monitor — Tue, 02 Oct 2012 21:02:50 GMT

AudioCodes announces that its Enterprise Session Border Controller product family has been validated by the ShoreTel Innovation Network, allowing deployment of ShoreTel Unified Communications systems with cost-saving SIP Trunking services.

AudioCodes’ family of Enterprise Session Border Controllers represents a key component for Service Providers and Businesses looking to migrate to a Voice-over-IP-based communications infrastructure. Located at the business premises, the E-SBC acts as the point of demarcation between the business’s VoIP network and Service Providers’ SIP-based services. The AudioCodes Mediant E-SBC product family includes four E-SBC hardware based platforms (Mediant 800, 1000, 3000 and 4000) offering a solution that covers organizations of any size and location, from small businesses and branch offices, to very large enterprise data centers and contact center facilities.

AudioCodes is showing the line of ShoreTel-validated Mediant E-SBC products in booth #808 at ITExpo, held October 2nd-5th, 2012 in Austin, TX and at the ShoreTel Champion Partner Conference, in Orlando, FL during November 7th-9th, 2012.

Vera Networks Launches Cloud-Based VoIP Call Flow Diagnostics Tool vCapture

VoIP Monitor — Mon, 30 Jul 2012 21:07:36 GMT

Vera Networks announces the availability of its vCapture service that enables operators of VoIP networks and platforms to quickly locate and help resolve call issues.

The vCapture service provides technical teams with the ability to target call issue-related SIP signaling data with laser-point accuracy without being in critically sensitive areas of the network infrastructure. The non-invasive, cloud-based, SIP signal capture technology records all SIP messaging within a switching network. This allows for in-depth analysis of session initiation protocol SIP signaling data, including call flow, signaling, and equipment-related problems.

The cloud-based solution, offered on a monthly subscription basis, reduces the time required to determine the root cause of most VoIP call and interop-related issues, improving customer response times, resource utilization, and reducing related operational expenses. It boasts an impressive array of features including an intuitive interface and protocol-aware searching and filtering capabilities.

Vera's vCapture service is the first cloud-based SIP capture solution and is capable of storing massive amounts of SIP signaling data, which then allows for retrieval of actual call information long after an event occurred.

The service was introduced in Chicago at International Telecoms Week earlier this year, and received enthusiastic reviews from the carrier community.

Comprehensive Study on SIP Trunking Reflects Emerging Enterprise Adoption

VoIP Monitor — Wed, 18 Jul 2012 20:51:52 GMT

Sonus Networks released the results of a global study which quantifies large enterprises deployment status, plans and attitudes about SIP Trunking and, by extension, Session Border Controllers. Commissioned by Sonus and conducted by Webtorials Editorial and Analyst Division, the “2012 SIP Trunking State-of-the-Market Report” found that while VoIP is a fully mature technology within corporate networks for intra-company communications, SIP Trunking, by contrast, is still in the early stages of deployment. In fact, roughly two-thirds of the respondents reported either “Significant Use” or “Extensive Use” of VoIP, while only about one-third of the respondents reported either “Significant Use” or “Extensive Use” of SIP Trunks. Among those using SIP Trunks, significant cost savings have been realized, with an average savings on the order of 33%.

Key Drivers: Direct Cost Savings and Ability to Add SIP-Based Features

The study featured responses from IT professionals at nearly 300 large enterprises. Among those respondents who have “Significant” or “Extensive” use of VoIP the study found that saving money is most important, but increased functions are also important.

68% of respondents indicated their decisions are driven “Mostly by cost savings” or “About equally” by cost and capabilities.
73% of the respondents indicated SIP Trunks and consolidation were a strong purchase driver.
The ability to “Add new SIP-based features” was a strong driver for 50% of the respondents.
VoIP (89%), Unified Communications (69%) and video conferencing (65%) are the most important types of media to be controlled via SIP.

The top inhibitor among “non-implementers” was not surprising, as 29% responded that services were not available at all of their network locations.

The Session Border Controller - An Essential Component of SIP Trunking

With the second generation of SBCs emerging, a number of additional features beyond security are now becoming available. Because of the critical role of SBCs in all types of communications, unrelated to security, “Reliability” ranks as the most important SBC feature for users at 88%. “Scalability” is a very strong second – at 78%, the high rank underscores that respondents understand multimedia communications mandate scalability for both multiple users and modes of communication. On a related note, 59% of respondents stated that their SBC must be “Capable of handling different types of multimedia traffic.” “Support for trunks with SIP and non-SIP traffic” was also significant at 58% as users clearly desire the capability to support both SIP and non-SIP traffic on the same trunk, allowing for dynamic utilization of all available bandwidth.

Grandstream IP Telephony Achieves Interoperability with Vitelity SIP Trunking

VoIP Monitor — Thu, 14 Jun 2012 21:38:19 GMT

Vitelity Communications and Grandstream Networks announce interoperability testing and certification of Grandstream's IP Telephony products with Vitelity's SIP trunking services has been successfully completed. Small- to medium-sized businesses and residential users adopting Vitelity's award-winning VoIP service for better financial savings, flexibility and enhanced productivity applications can seamlessly deploy Grandstream's desktop IP Telephony products, including all GXP IP Phones and GXV IP Multimedia Phones.

Vitelity Communications offers unique and flexible VoIP solutions for customers worldwide. "We are thrilled to offer our worldwide customers the opportunity to take full advantage of, and realize the extreme quality of Vitelity's VoIP service offerings by using Grandstream IP phones for the best possible user experience," says Kerry Garrison, Vice President of Strategic Initiatives at Vitelity. "Customers connect with the Grandstream brand -- with its value, reliability, and ease of use. We value this strategic partnership and look forward to jointly evolving our IP offerings to benefit the needs of our customers."

Grandstream GXP Series of Enterprise SIP telephones provides affordability, best-in-class HD audio, a comprehensive set of advanced call features, multi-language support, security protection, simplified management and automated provisioning with TR-069 or HTTPS, advanced XML customization and more. For customers seeking a more advanced, integrated multimedia experience with free real-time video calling capability and instant access to real-time web and social multimedia applications, Grandstream's award-winning GXV IP Multimedia Phones are available.

Ingate Debuts Full Lineup of Software SIParator and Firewall E-SBCs

VoIP Monitor — Tue, 05 Jun 2012 21:47:40 GMT

Ingate announces the Ingate Software SIParator and Ingate Software Firewall, new software-only versions of the company's Ingate SIParator and Ingate Firewall E-SBCs.

The Ingate Software SIParator/Firewall offers the same security and SIP-enabling functionality found in Ingate's hardware-based E-SBCs. The software can be installed on customers' own servers (or integrated with IP-PBXs or media gateways) or used as a virtualized application and all of Ingate's usual advanced modules can be added as required.

Available now, the Ingate Software SIParator/Firewall comes in a wide range of models to address the needs of small businesses and SMBs, large enterprises and everything in-between. The software-only E-SBC can handle from as few as five simultaneous calls, up to as many as 10,000, depending on the hardware used.

The software-only E-SBC is intended for IP-PBX vendors, system integrators and customers deploying a large number of Ingate products on their own hardware platform.

Enabling Secure SIP Trunking, Unified Communications
Like all Ingate E-SBCs, the Ingate Software SIParator/Firewall enables secure SIP into the network to make trusted SIP trunking and UC possible. It works hand-in-hand with an existing network firewall to allow SIP traffic to traverse the enterprise edge. It can also be configured with firewalling functionality to provide enterprise security for all SIP and data traffic.

Advanced Security Bundled Free with Ingate Products
Intrusion Detection System/Intrusion Prevention System solutions for SIP are bundled free with all Ingate models, including the software version of the Ingate E-SBC. IDS/IPS protects against attacks targeting SIP devices, such as IP-PBXs and SIP phones. IDS/IPS works in tandem with Ingate's existing security features to offer maximum protection.

snom Offers Free SIP Online Training Course for Resellers

VoIP Monitor — Fri, 01 Jun 2012 16:00:55 GMT

snom announces a new e-learning course focused on SIP fundamentals - SIP 101. This course is the latest addition to snom's new online training program introduced earlier this spring and is available free to registered snom VARs from North and South America, Australia and New Zealand. SIP 101 will serve as a prerequisite to enrollment in the company's online snom ONE Certification Training course.

Busy VARs can gain valuable, hands-on experience with all aspects of SIP, without the time and expense of traditional courses. "By enrolling in our SIP 101 e-learning course VARs from around the world can learn fundamental SIP basics online and on their own time. This course is a prerequisite first step for VARs on the path to our snom ONE Certification Training and new business opportunities created by working with the snom ONE family of IP PBX solutions," said Mike Storella, chief operating officer of snom technology, Inc.

The SIP 101 course is a modularized, online, Flash-based training course, complete with animations and quizzes to help engage trainees. Participants can complete an online assessment upon course completion to earn snom SIP 101 certification.

Eligibility for the SIP 101 certification class is open to registered snom VARs in the Americas, Australia and New Zealand who can register quickly and easily at onlinetraining.snom.com. snom will be offering more online courses about snom products and key aspects of its SIP-based technology in the coming months.

Vitelity Launches SIP Enable Communications Solution

VoIP Monitor — Mon, 14 May 2012 20:39:38 GMT

Vitelity Communications announces the launch of the communications bundle, SIP Enable, at the International Telecoms Week conference. Vitelity partnered with Patton Electronics to develop the product, which enables legacy phone systems that do not have native SIP capabilities to now be able to use SIP trunk services from Vitelity. The systems can integrate the advanced technology into the system in order connect to the Internet and utilize additional features, such as SMS, vFax and global DID origination.

For those with legacy PBX systems, SIP Enable with Vitelity SIP service is the easiest way to increase functionality and take advantage of competitive calling plans, according to Tyler Delin, SmartNode Product Manager at Patton Electronics.

Stop by the Vitelity Communications booth (#1513) at the International Telecom Week conference to learn more about the SIP Enable product and enter to win a free SIP Enable bundle.

WIND Mobile Selects Acme Packet for SIP Interconnect

VoIP Monitor — Tue, 08 May 2012 20:51:06 GMT

Acme Packet announces that WIND Mobile selected the Acme Packet Net-Net Session Director session border controllers to securely connect WIND's VoIP core network with other service providers. The Acme Packet Net-Net Session Director enables a secure, flexible and interoperable SIP interconnect solution for off-net voice traffic, helping reduce costly origination and termination time-division multiplexing fees.

Using SIP interconnect, WIND's calls travel to other service providers over IP, which eliminates costly TDM fees and scales the company's VoIP interconnect relationships. The Acme Packet Net-SAFE's security architecture protects WIND from denial of service attacks and network overloads with access control lists, topology hiding and dynamic rate limiting.

The Net-Net Session Director's flexible routing capabilities helps WIND reduce session expenditures while optimizing subscriber experience. Acme Packet's interworking capabilities simplifies WIND's network operations, assures multi-vendor interoperability and accelerates the company's time-to-market.

Acme Packet's session border controller capabilities also include high availability, call admission control and quality of service capabilities to ensure network reliability. Also, with Acme Packet's feature rich and standards-based solution, WIND's network will be able to utilize key encryption, transcoding and interworking capabilities in the future.

AudioCodes Executives to Speak on Enterprise Connect SIP Trunking Tour

VoIP Monitor — Tue, 01 May 2012 22:19:41 GMT

AudioCodes announces that two of its executives plan to speak on the Enterprise Connect SIP Trunking Tour, a four-city tour promoting SIP Trunking services, education and technologies. As part of the event series, Alan D. Percy, Senior Director of Marketing for North America and Larry Clarkson, Chief Technology Office for North America will participate in panel discussions, offering real-world customer experiences, best practice advice, and tips when migrating from legacy TDM trunking to SIP Trunking services.

SIP Trunks are becoming more widely available and have tremendous potential—they can reduce enterprise costs for carrier services, as well as getting your enterprise one step closer to supporting true Next-Generation IP Communications. In response to the booming demand for SIP Trunks—and for information about SIP Trunks—Enterprise Connect is launching a four-city “road show” on this vital topic.

Enterprise Connect SIP Trunking Tour Locations and Dates:

May 9 :: Las Vegas :: Mandalay Bay
May 16 :: New York City :: American Conference Centers
May 22 :: San Francisco :: Grand Hyatt
June 6 :: Chicago :: Hyatt Regency

The program will feature an intensive day-long series of sessions and networking opportunities designed to help enterprise decision-makers understand the market for SIP Trunking and the technologies that drive it. That includes the following key topics:

SIP Trunking ROI – What We Really Know?
Technical Challenges – What’s Needed to Make SIP Trunks Work?
What’s SIP Trunking’s Role in Overall Architecture?
Do’s and Don’ts for SIP Trunking – Case Studies

In addition to these information-packed sessions, attendees be able to bolster their understanding of SIP Trunking by meeting informally with other colleagues at the tour event, during networking breaks, lunchtime, and an end-of-day cocktail reception. Attendees will also have the chance to meet with a number of AudioCodes executives and channel partners.

More information and registration for the event can be found here.

A special registration discount is available for pre-registration using the discount code “AUDIOCODES”

Groundwire iOS SIP Client is Now Available on Android

VoIP Monitor — Mon, 30 Apr 2012 21:44:50 GMT

Acrobits announces the long awaited release of their popular business caliber SIP client Groundwire on Android. Groundwire includes all the features of Acrobits Softphone as well as the additional features business users need. Including but not limited to transfer and attended transfer, call conferencing, multi line and voicemail notification; Groundwire puts all the tools professional SIP users need in the palm of your hand.

With Groundwire, Acrobits also brings support for ZRTP to Android. The most advanced method for call encryption in VoIP, ZRTP is a must have for users who want the most secure calls possible. In addition, Acrobits adds support for SDES SRTP. Both features will be available in both Acrobits Softphone and Groundwire for Android. Groundwire is available on Google Play and the Amazon Marketplace now. A new update for Acrobits Softphone is also available which adds support for ZRTP and SDES SRTP.

And expect more from Acrobits soon. They are now working on an iPad specific version of their iOS clients, which Acrobits promises will be the first VoIP Client truly designed for the tablet experience.

Digium Simplifies Communications With Advanced Asterisk-based VoIP Gateways

VoIP Monitor — Mon, 26 Mar 2012 21:13:40 GMT

Digium introduces the G100 and G200, the first in a family of cost-effective VoIP gateways that simplify the process of deploying converged media networks. Built on a powerful combination of the Asterisk open source communications engine and a state-of-the-art embedded platform, the new gateways provide the best value for Asterisk communications solutions.

Digium’s gateways are built to support both TDM-to-SIP and SIP-to-TDM applications. In a TDM-to-SIP deployment, the gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.

The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI interface, which allows for easy navigation and effortless setup. The gateways feature a power-saving embedded design with a highly efficient digital signal processor handling all media-related operations. The combination of an intuitive user interface, the flexibility of Asterisk and the purpose-built media processing capabilities of the DSP results in a gateway platform that outperforms the dated designs in the market today.

Digium beta testers agree. “Setting up the G200 was extremely easy compared to doing it with other gateways. I'm spoiled now!” said Tim Banks of Project Resource Solutions, an Illinois-based Digium Select partner. “Digium has really set the bar high. Their new gateways make it incredibly easy to connect older TDM phone systems with SIP services.”

Digium’s new gateways represent a solution to one of the challenges associated with running Asterisk applications in virtualized environments. TDM interface cards require a card slot – something distinctly missing from virtual servers. By converting the media and signaling from TDM to SIP on a dedicated external device, Asterisk users can migrate applications to virtualized, hosted or cloud environments.

The G100 includes a single software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls. The G200 doubles the capacity with two T1/E1/PRI interfaces and up to 60 concurrent calls. Both models have integrated echo cancellation, a small footprint (1U, half-width, half-depth) and no failure-prone moving parts.

The single-span G100 lists for $1,195 USD while the dual-span G200 model lists for $1,995 USD. The gateways are currently available worldwide through Digium’s network of distribution and integration partners.

Dialogic Solution Enables SIPxchange to Provide Carrier-Grade VoIP Service

VoIP Monitor — Tue, 13 Mar 2012 21:01:21 GMT

Dialogic announces that SIPxchange chose Dialogic as its core technology provider. The Dialogic ControlSwitch System and media gateway solutions provide SIPxchange with robust routing capabilities, while supporting both TDM and IP traffic. This positions SIPxchange to reliably and cost-effectively support its rapidly expanding DID origination, Toll-Free and VoIP Termination network that provides ubiquitous Local Access Transport Area coverage in the key markets of Texas.

In need of a reliable solution that would support its existing customer base while allowing for growth throughout the United States, SIPxchange chose Dialogic for its strong leadership position in class 4 switching, support for SS7, its carrier-grade Dialogic I-Gate 4000 PRO Media Gateway, and an extensive array of Codecs and Protocols, including SIP, MGCP, H.323, T.38, G.711, G.729, G.723, GSM. In addition, as VoIP becomes the emerging choice for business among carriers within the United States, SIPxchange needed a solution that would allow for cost-effective direct interconnection to the Legacy PSTN Tandems. Dialogic's ControlSwitch System, with its distributed media gateway architecture, provides reliability and scalability for interconnection to the PSTN.

Dialogic's ControlSwitch System and media gateway solutions require very little in the way of hardware equipment, and as a result, SIPxchange was able to reduce its number of servers from 17 to four, thereby further reducing its capital and operational expenditures and moving toward a more eco-friendly business model.

Grandstream Introduces New GXP Enterprise HD Telephone

VoIP Monitor — Mon, 05 Mar 2012 21:08:16 GMT

Grandstream Networks introduces the GXP2124 Enterprise HD IP Telephone for enterprise customers looking for a high performance HD telephone with numerous programmable keys and Electronic Hook Switch support for high call volume applications such as call centers, customer support and reception areas.

The GXP2124 is Grandstream’s first HD IP telephone with EHS support for Plantronics headsets. Users can answer and end calls using only the button on the headset – eliminating the need to touch the desktop phone. In addition, the GXP2124 features 4 line keys with up to 4 SIP accounts, 24+4 programmable speed-dial/BLF keys, broad interoperability with major SIP platforms such as Broadsoft/Asterisk/etc, superior HD audio, and 5-way conference. Grandstream is showcasing the GXP2124 and the entire family of award-winning GXP Enterprise IP Telephones at Stand B76, Hall 13 at CeBIT being held this week in Hanover, Germany.

Key Features of Grandstream’s GXP2124 Desktop HD Telephone:

4 lines with up to 4 SIP accounts, 24+4 speed-dial/BLF extension keys with dual-color LED, 4+4 context sensitive XML programmable keys, up to 32 call appearances, support for Electronic Hook Switch with Plantronics headsets
240x120 backlit graphical LCD with up to 8 level grayscale, dual 10M/100Mbps Ethernet ports with integrated PoE
Support for multiple native languages (including English, German, French, Spanish, Italian, Portuguese, Chinese, Korean, Japanese, Russian, and Greek)
HD wideband audio, full-duplex high performance hands-free speakerphone with advanced acoustic echo cancellation, 5-way conference by leveraging the superb audio performance of DSP Group’s XciteR chipset.
Integrated real-time web applications (weather, stock, currency, etc.), large phonebook (up to 2,000 contacts) and call history (up to 500 records)
Advanced security protection (TLS/SRTP/HTTPS/802.1x) and auto provisioning (TR-069, HTTPS, and AES encrypted XML configuration file)

Pricing and Availability

The GXP2124 will be generally available for purchase by the end of March 2012 through Grandstream’s distribution channels at a list price of US$169 (North America). For other regions, please contact your local Grandstream distributors/resellers.

Revolabs Unveils FLX VoIP

VoIP Monitor — Tue, 31 Jan 2012 22:02:34 GMT

Revolabs introduces the Revolabs FLX VoIP, the first wireless conference phone designed for VoIP networks. Supporting a wide variety of IP switches, the FLX VoIP is the only conference phone that supports the audio clarity of HD audio while providing the freedom of wireless microphones and speakers. The feature set that has been available through the Revolabs FLX for analog phone lines is now also available for IP telephone networks, providing unprecedented conference call clarity and flexibility.

The FLX VoIP integrates directly with most IP telephone switches following the SIP standard. Through this integration, new features only available through digital switch environments, such as voice mail alerts and "do not disturb," can now be offered with the FLX VoIP. The phone's wireless capabilities allow it to be used in small and midsize conference rooms without running any cables. As with the FLX for analog phone lines, this allows for a clean look while requiring less space on the conference table. The independent microphones, speaker, and dialer of the FLX VoIP give the user freedom and flexibility that other conference phone systems cannot offer.

Combining wireless operation, high-quality wideband audio, 128-bit encryption, and integrated Bluetooth®, the FLX VoIP redefines the conference speakerphone. Unlike the single-component design of previous solutions, Revolabs FLX VoIP evolves the conference phone into several distinct components, giving users unprecedented freedom with respect to placement and accessibility of the speaker, microphones, and dial pad.

Available with a variety of compatible Revolabs microphones, the FLX VoIP supports a lapel microphone worn by one person; an omnidirectional tabletop microphone that captures the sound of six to 10 participants; and a directional tabletop microphone that enables audio capture from two to three people. Because the FLX VoIP dialer operates like a telephone for handset calls and enables the set up of conference calls, there is no need for a separate desk and conference phone.

The Revolabs FLX VoIP can also serve as the audio interface for virtually any major brand of video conferencing equipment, making it the ideal unified communication technology for small to medium-sized conference rooms, executive offices, and small office/home office environments. FLX VoIP's integrated Bluetooth technology provides a single collaboration device no matter which communication channel is used, allowing users to connect speakers and microphones to their Bluetooth-enabled mobile phones or computers.

The Revolabs FLX VoIP will be available worldwide in February 2012, and sold through major distributors, dealers, and resellers.

Acrobits’ Video VoIP-SIP iPhone Application Challenges Skype and Apple’s Facetime

VoIP Monitor — Tue, 24 Jan 2012 22:03:52 GMT

Acrobits released its new combined Voice and Video application with SmoothFlow video technology for Apple’s iPhone.

The Voice and Video applications are available in two versions; one for the consumer market (SIP Phone) and one in a fully featured business phone application (Groundwire) with both apps loaded with features such as HD Sound, Phone book integration and Avatar Dialing. Consumers can choose from hundreds of pre-configured VoIP carriers across the world while the Groundwire business application works with both closed and open source office IP-PBX platforms such as Cisco, Avaya and Asterisk.

Global VoIP Training School Chooses OnSIP as SIP Provider

VoIP Monitor — Mon, 09 Jan 2012 23:39:41 GMT

As the provider of the global standard in SIP training and certification, The SIP School has taught thousands of employees in the telecommunications industry how to better support their clients, products, and services. Until recently, students training to become an SIP School Certified Associate were instructed in their first session to create a SIP address with any free service. Today, The SIP School announces another option by working with OnSIP as their SIP service provider – leveraging the OnSIP API to provision each student with a SIP address on thesipschool.com domain.

OnSIP originally began SIP domain hosting to encourage their customers to simplify communications and boost their corporate branding by creating SIP addresses for employees that match their email addresses. With the OnSIP API, customers can integrate SIP address provisioning into their own service offerings as The SIP School has accomplished.

Telchemy Announces Powerful New Active Test Tool for VoIP, Videoconferencing and Network Test

VoIP Monitor — Fri, 18 Nov 2011 17:45:52 GMT

Telchemy announces DVQattest Version 2.0, a powerful new active test system for unified communications. DVQattest generates Voice over IP and Videoconferencing calls with SIP signaling, synthetic HTTP, POP3, SMTP, DNS and DHCP transactions and a range of IP network tests. This advanced test product supports pre-deployment testing, SLA monitoring and troubleshooting for converged networks and services.

DVQattest Agents are compact but highly featured software applications that can be installed on a range of operating systems and hardware platforms, including Linux servers and appliances, Android mobile phones and directly integrated into network equipment and CPE. Tests can be run on-demand or scheduled to run indefinitely. Agents can run multiple concurrent tests to other Agents or to IP phones, Web sites, Email sites and other network-based services. DVQattest Agents support complex networks with overlapping IP address spaces, VLANs and a range of SIP infrastructure configurations.

VoIP and Videoconferencing tests verify the performance of both SIP signaling and the media stream. Voice and Video quality is measured using Telchemy’s market-leading VQmon technology, providing MOS scores and a wide range of diagnostic data. Voice over IP tests support configurable codec, packet size and jitter buffer configuration; Videoconferencing tests support configuration of codec, image size, GoP, frame rate, bit rate and smoothing.

Network tests include Agent-to-Agent tests that measure loss, jitter and available bandwidth in each direction, industry standard network tests and advanced route diagnostics. DHCP and DNS tests verify correct operation of vital network functions and HTTP/POP3/SMTP tests measure performance of key applications.

The DVQattest Controller provides an easy-to-use management application that supports test definition, remote DVQattest Agent management and test result collection and reporting. For larger networks, the SQmediator performance management application provides a scalable and intuitive solution for multiple concurrent users. DVQattest Controller and SQmediator support key security requirements and maintain AES encrypted connections to DVQattest Agents.

DVQattest provides dependable, accurate and detailed performance metrics and has already been deployed in critical network applications. When the US Department of Defense needed accurate tools for measuring performance for Internet Routing in Space (IRIS) satellite based router project, DVQattest provided a key element of their measurement infrastructure.

DVQattest is available in a wide range of configurations, suitable for mid-large enterprise, hosted and cloud based services and tier one service providers.

SIP Forum Announces Dates for Second Annual SIPNOC

VoIP Monitor — Wed, 09 Nov 2011 22:32:10 GMT

The SIP Forum announced today that it will host the second annual SIP Network Operators Conference, a two day educational conference focusing on the challenges and opportunities related to the deployment of SIP-based carrier services globally. SIPNOC US 2012 will be held in Herndon, VA on June 25-27, 2012 and will build on the success of the inaugural event last spring, which attracted leading technical and operations personnel from the global carrier community and earned high praise from attendees for its educational, non-commercial and technical content that focused on the real-world challenges operators face when deploying SIP services in global IP networks.

The SIP Forum will release further details about SIPNOC US 2012 including sign-up for early bird registration in mid-November at its conference website http://www.sipnoc.org.

SIPNOC 2012 will bring together leading technical minds from the telecommunications industry to learn, discuss and formulate new ideas and strategies concerning the challenges and opportunities for SIP-based carrier services in fixed and mobile IP network environments. The conference, which is designed specifically for SIP network operations personnel and engineering staff, will feature well-known industry speakers and a number of highly technical educational and instructional panels and sessions. The SIP Forum will also host networking events at the conference and offer a series of informal “Birds of a Feather” sessions, which encourage discussion of varying topics held in hallways, available meeting rooms and break-out areas.

SIPNOC 2012 will build on themes first discussed at last year’s inaugural event: addressing issues critical to the reliable and successful deployment and operation of SIP-based services in carrier networks and the opportunities that come with it. Among the topics expected to be on the agenda at this year’s conference are SIP trunking interoperability and the SIP Forum’s recently ratified SIPconnect 1.1 technical specification that provides a definitive and standardized set of guidelines for seamless, end-to-end interoperability between SIP-enabled IP-PBXs and service provider networks. Other themes to be discussed include Fax over Internet protocol interworking, implementing SIP with IPv6, and the sharing of best practices for the utilization of Wireshark for network diagnostics within IP network environments.

Building on the success of these US-based SIPNOC events, the SIP Forum is expected to announce details for a Europe-based event - SIPNOC EU 2012. The dates and location of the next event will be released later this year.

SIPNOC US 2012 will be held at the Hyatt Dulles Hotel in Herndon, VA June 25 -27, 2012.

snom Unveils New Class of SIP Phones Designed for SMBs with Big Business Tastes

VoIP Monitor — Mon, 07 Nov 2011 22:03:52 GMT

snom introduces a new line of business VoIP phones – the snom 7xx series designed for both small and mid-sized businesses requiring an enterprise-class desktop phone on an SMB budget. The snom 720 and snom 760 business phones bring together the multiple programmable buttons and popular standard business functionality of the snom 3xx series with the advanced functionality, sleek styling and Gigabit Ethernet switch found in the snom 8xx series to create an advanced desktop phone at a value-driven price.

Advanced Features and Elegant Design for Next Generation Business

Both the snom 720 and 760 offer a Gigabit Ethernet switch, automatic provisioning, wireless LAN connectivity and snom’s superior wideband high definition voice quality. In addition, thanks to a Gigabit Ethernet switch, both phones can transfer data at a speed of 1000Mbits/s without slowing down the network or a connected PC. Both phones also feature Bluetooth connectivity via optional USB stick, allowing users the freedom to use a compatible Bluetooth headset with their snom 7xx series phone. The snom 760 features a high-resolution color display and two USB ports for a variety of connectivity options, as well as a newly designed handset grip that increases user friendliness by providing silent pickup and return of the handset. The snom 760 also includes a 16-key programmable busy lamp field and 4 context-sensitive keys complemented by the large, easy to read display.

The snom 760 also offers the standard desktop feature set of any snom phone, and is ideal for business environments that require a greater level of visual functions, such as the use and delivery of XML-based data. The large display also supports caller images, uploaded by the caller or included in the user’s address book.

Traditional Phone Features for Everyday Business

The snom 720 builds off the elegant and functional simplicity of the snom 3xx series business phone, featuring an easy to read, four-line monochrome graphical display. The snom 720 offers 18 fully configurable function keys and four variable keys, ideal for managing and contacting large groups of people.

The snom 720 also supports all standard VoIP calling features, including an address book with 1,000 possible entries, speed dialing, URL dialing, ringtone selection and LED call indication. In addition, the snom 720 and 760 also feature wireless LAN (WLAN) connectivity via optional USB stick.

Both the snom 720 and snom 760 are available for order today by distributors and resellers worldwide. snom 720 MSRP is $219.00 US and snom 760 MSRP is $329.00 US.

Panasonic Announces Interoperability for Full Lineup of SIP Phones With CoreDial

VoIP Monitor — Thu, 03 Nov 2011 21:16:31 GMT

Panasonic announces that CoreDial has certified Panasonic's line of SIP Phones, the KX-TGP500, KX-TGP550, KX-UT113-B, KX-UT123-B, KX-UT133-B and KX-UT136-B, for use with their Hosted PBX telephony platform. This alliance leverages Panasonic's global leadership in the DECT cordless telephone market and CoreDial's leading private label VoIP cloud software platform, adding up to a winning combination.

Ideal for both home office and business environments, Panasonic's SIP phone systems offer the flexibility of cordless or corded models while supporting a wide range of business class features provided by the CoreDial platform. The KX-TGP500/550 systems offer convenient, cordless designs that eliminate the need to run dedicated network wiring to each employee work station while incorporating DECT 6.0 to ensure no interference with wireless networks. The new KX-UT Series is designed to complement a company's existing communication infrastructure and offer end-user savings with features including two data ports, PoE and lower power consumption while in ECO mode. All Panasonic SIP models are HD Voice enabled, allowing for outstanding voice quality, and offer flexible system expandability.

With flexible configuration options, it has never been easier to deploy and expand a Panasonic SIP-based phone system with CoreDial's Hosted PBX platform. The reduced hardware costs and simplicity of routing calls over an Internet connection can add up to huge savings on monthly telephone bills, thus enabling all business environments to take advantage of a larger variety of business-class features such as call forwarding, intercom and conferencing, voicemail and more.

TGP500 Series Details:

KX-TGP500: The system features a wall-mountable base unit and one cordless handset. It is expandable up to 6 DECT 6.0 cordless handsets and supports up to 8 phone numbers and 3 simultaneous calls. It boasts Wide Band Audio (G.722) and 5 hours Talk Time, 10 days Standby. Its elegant design features a white backlit large LCD on the handset and a Handset locator button on the base unit. It also has a handset Speakerphone, 2.5mm headset jack and belt clip.

KX-TGP550: The KX-TGP550 has all the features and benefits of the KX-TGP500 and adds a corded base unit with a large white backlit LCD and 5 hours Talk Time, 10 days Standby, plus a Hands-Free Speakerphone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication.

KX-TPA50: The TGP500 systems can be expanded up to a total of 6 cordless handsets by adding the KX-TPA50 cordless handset.

KX-UT136 and KX-UT133: These models are user-friendly and easy to operate, with 24 programmable feature/functionality keys. The KX-UT136-B features a six-line backlit graphical LCD and 2 Ethernet ports, while the KX-UT133-B offers a three-line backlit graphical LCD and 2 Ethernet ports.

KX-UT123 and KX-UT113: These standard models are breaking barriers by offering HD Voice, PoE and two-year warranty. The UT123 features a three-line backlit graphical LCD and two Ethernet ports, while the UT113-B has a three-line graphical LCD and one Ethernet port. They both offer ease-of-use at a competitive price for excellent return on investment.

Panasonic Showcases Award Winning SIP Telephony Solutions at AstriCon 2011

VoIP Monitor — Tue, 25 Oct 2011 21:25:13 GMT

Panasonic is showcasing an extensive range of IP telephony solutions at AstriCon in Panasonic booth #204 at the Westin Westminster in Denver, Colorado, October 25-27.

As an AstriCon 2011 Gold sponsor, Panasonic is demonstrating a broad range of SIP telephones including SIP Cordless Phone Systems and KX-UT series which are interoperable with Asterisk's open source PBX. Panasonic's SIP telephones work with the Asterisk platform which offers both classical PBX functionality and advanced UC features.

In its seventh year, AstriCon is the longest running conference devoted to the Digium Asterisk communications platform. AstriCon brings together open source enthusiasts, from coders and system integrators to service providers and enterprise IT professionals, who are looking for an in-depth understanding of Asterisk open source technology.

Panasonic's SIP Phone Systems:

The Panasonic SIP Cordless Phone System is a small business communication solution that offers the flexibility of convenient expansion as a company grows. The KX-TGP500 system features a wall-mountable base unit and one cordless handset. Expandable up to six DECT 6.0 cordless handsets, the model supports up to eight phone numbers and three simultaneous calls. It boasts Wide Band Audio (G.722) and five hours Talk Time, 10 days Standby. Its elegant design features a white backlit large LCD on the handset and a Handset locator button on the base unit. It also has a handset speakerphone, 2.5mm headset jack and belt clip.

The KX-TGP550 model has all the features and benefits of the KX-TGP500 and adds a corded base unit with a large white backlit LCD, plus a Hands-Free Speaker phone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication.

Also on display, the Panasonic KX-UT series offers a cost-effective communications solution for businesses of all sizes that leverages the latest developments in Hosted and Open Source PBX technologies and is designed to complement a company's existing communication infrastructure. Most models feature two data ports so users can connect a second network device without the time and expense of running an additional Ethernet cable. The KX-UT series models are Power over Ethernet ready which eliminates the need for additional electrical adaptors. Wide-band, high-definition audio (G.722 codec) coupled with echo cancellation and an expanded acoustic chamber allows the KX-UT series to offer crisp sound quality for crystal clear conversation.

Panasonic is also previewing the new KX-UT670, a highly expandable corded SIP phone with a seven-inch color LCD touch screen function that will help to transform business communication. Additional key features include HD Voice (G.722), two Ethernet ports, 3-way conference calling, IP camera integration, full duplex speakerphone, 100 entry phonebook and PoE ready.

4PSA Enhances VoIP Suite with Cloud Telephony Service

VoIP Monitor — Fri, 14 Oct 2011 21:45:01 GMT

4PSA announces the public availability of Cloud Telephony, the flexible, next-generation SIP trunking service that can be provisioned within minutes.

The new service delivers access to the telephony network using the SIP protocol and features unlimited concurrent incoming and outgoing calls. Moreover, it is possible to choose local phone numbers in over 30 countries around the world with best rates for domestic and international calls. Using VoIP technologies and the Cloud Telephony service, it is not necessary to install physical phone lines and incoming calls are always free.

new service is part of 4PSA's strategy to bring the cloud flexibility to resources that traditionally required complicated provisioning processes. Cloud Telephony follows cloud Unified Communications and cloud software licensing as a way to simplify the deployment of traditional telephony services.

Cloud Telephony can be used with any communication system that implements a VoIP SIP interface. When paired with VoipNow Cloud Instance that already offers software, infrastructure and support, it is a complete solution for service providers so that they can start delivering services to their customers immediately, and also for businesses that are looking for a turn-key Unified Communications solution.

Network Equipment Technologies Gains SIP Trunking Certification With AAPT

VoIP Monitor — Mon, 03 Oct 2011 22:41:17 GMT

Network Equipment Technologies announced that its UX Series with Session Border Controller, a qualified gateway for Microsoft Lync Server 2010, is now certified to work with AAPT's SIP trunking service. AAPT is 100% owned by Telecom New Zealand. The company's SIP voice service is a SIP trunking solution, allowing customers with an IP-enabled PBX or SIP gateway device to connect to AAPT via Ethernet and have their telephony traffic carried via IP utilizing SIP, providing a far more scaleable alternative to traditional ISDN.

Qualification testing of NET's UX Series with AAPT's SIP trunking was performed through a collaborative effort between AAPT's interoperability team and NET. Customers seeking to integrate legacy PBX and IP-PBX solutions and deploying Microsoft's Lync Server 2010 can now use NET's UX Series as the enterprise SBC and gateway for the termination of AAPT's SIP trunking services and/or co-located E1 ISDN services. This solution provides enterprise customers with the flexibility and cost-savings associated with SIP trunking combined with the productivity gains of unified communications on the Microsoft Lync 2010 platform.

The UX Series, in an enterprise-SBC role, provides high-performance VoIP communications for enterprise Microsoft Lync deployments with AAPT. The UX Series routes SIP sessions; converts signaling and media between Microsoft Lync and the AAPT SIP trunk; and acts as a demarcation device between the enterprise network and the AAPT network for establishing a reliable communication service.

Trisys Introduces Replay SIP

VoIP Monitor — Fri, 09 Sep 2011 21:16:30 GMT

Trisys introduces Replay SIP, a scalable module of its popular Replay Call Recording solution that is easily added to IP-based telephony systems. As business adds IP phones to existing telephony systems in order to take advantage of cost efficiencies, access to phone application software is often lost or requires expensive upgrade. Now with Replay SIP business can freely add call recording functionality for less than $300 per phone. The small footprint, scalable product also moves Replay to the forefront of options for new, predominantly IP phone system sales.

Trisys’ Replay SIP is a 100% software based call recording solution. It is designed to record phone conversations taking place on SIP-based IP phone systems. Replay SIP runs unobtrusively on networks, monitoring VoIP traffic for desired calls, and converts them in to call recordings. With Replay SIP installed, authorized users can easily access call recordings for quality assurance, regulatory compliance, dispute resolution, and much more.

Replay SIP supports most SIP-based IP telephone systems, saves recordings as standard WAV files, which can be automatically archived or deleted, supports On-Demand and Pause/Resume recording providing that PBX and IP Phones support RTP “events” as per RFC 2833/4733. Replay SIP is available as a software "only" or as a complete turnkey solution including software and hardware.

snom Partners with SIP Print to Provide Interoperable and Integrated SIP Call Recording Capabilities

VoIP Monitor — Tue, 26 Jul 2011 20:23:12 GMT

snom has forged a new technology partnership with SIP Print, a maker of SIP-based call recording, accounting and management software. After a battery of tests, SIP Print’s call recording platform has been approved as interoperable with snom’s portfolio of desktop phones and IP PBXs, including the snom ONE plus IP PBX appliance.

With both companies well-entrenched in the small and medium-sized business space, the partnership offers smaller companies and organizations an enterprise-class solution for call recording and accounting in a package designed specifically for SMBs. For organizations either required to track and record calls for compliance reasons, such as emergency call centers or small financial or legal institutions, or interested in call recording for internal business intelligence purposes, the snom-SIP Print combination provides an easily installed and managed, end-to-end solution with minimal capital investment.

Based on standards-based SIP technology, the snom ONE IP PBX is a versatile VoIP communications platform for SMBs with five to 150 extensions or more. The snom ONE offers all the functionality available with VoIP telephony, including call waiting, call forwarding, centralized address book, conferencing and simultaneous ringing of cell phones and desktop phones. In addition, it offers advanced features such as PSTN or SIP trunk connectivity, shared line emulation, hot desking and presence features. The snom ONE plus allows businesses to put all of the features and functionality of the snom ONE IP PBX in an on-premise hardware solution.

With snom’s suite of award-winning desktop phones and endpoints, such as the snom 3xx series desktop phone, the full-color touchscreen snom 870, the snom m9 wireless DECT phone and related endpoints, such as the MeetingPoint conference phone, the combination provides an end-to-end VoIP platform purpose-built SMBs and emerging enterprises.

SIP Print’s call recording appliance comes in various sizes, ranging from 15 seats to more than 200 seats. It provides a searchable database of calls, archived in .wav format for easy control of playback and integration, and is CALEA compliant.

Integra Telecom Earns SIP Interoperability Certification with Digium’s Switchvox SMB

VoIP Monitor — Tue, 12 Jul 2011 21:54:53 GMT

Integra Telecom has earned Digium’s SIP Trunking Interoperability certification with Digium’s Switchvox SMB unified communications solution.

This certification enables the Digium UC solution to work seamlessly with Integra’s SIP Trunking. Integra is expanding the number of business phone systems certified to operate with its SIP Trunking service so its business customers can benefit in the following ways:

Gain more efficient use of bandwidth by combining voice and data networks into a single service
Receive dynamic bandwidth allocation so that voice is prioritized over data, meaning Internet bandwidth will increase when voice sessions are not in use
Maximize capacity through multiple compression options that place more calls over the same bandwidth with no need for additional circuits

Integra will support Digium’s Switchvox SMB 65, 305 and 355 models.

Through Integra Telecom, businesses can utilize Internet, MPLS, SIP services and other voice lines integrated with state-of-the-art VoIP systems, data services and unified communication applications.

SIP Forum Announces Ratification of Version 1.1 of the SIPconnect Technical Recommendation

VoIP Monitor — Tue, 17 May 2011 17:02:28 GMT

The SIP Forum announces that it has ratified Version 1.1 of the SIPconnect Technical Recommendation, with the unanimous approval of the SIP Forum Board of Directors. The new version of the recommendation, developed by the SIP Forum’s SIPconnect Task Group, is a follow-on to Version 1.0 ratified in 2008, and provides a more definitive and standardized set of guidelines for seamless, end-to-end interoperability between SIP-enabled IP-PBXs and service provider networks.

The SIPconnect Technical Recommendation, which represents the consensus of a broad cross section of the global telecom community, is one of the most important initiatives of the SIP Forum, aimed at providing an up-to-date, international framework for direct IP peering between SIP-enabled enterprises and service provider networks, ensuring the interoperability of network elements across the IP environment and providing a level playing field for vendors and service providers as they develop new equipment and IP applications for deployment.

The SIPconnect 1.1 Technical Recommendation features an array of enhancements from Version 1.0 such as more comprehensive guidelines about security and SIP end-point and media endpoint functionality. Highlights include:

Standards-based support for both Static (DNS-based) and Registration (SIP REGISTER-based) modes of operation incorporating the newly approved RFC 6140
Description of SIP endpoint functionality required for interworking, with detailed discussion of various error conditions and appropriate responses to those errors
Description of media endpoint functionality required for interworking
Focus on Phone number (i.e., E-164) based SIP Address of Record
Additional voice services using SIP techniques
A detailed description of transaction layer security usage
A roadmap on what implementers can expect in subsequent SIPconnect revisions (IPv6, emergency services, etc.)

SIPconnect puts forth a technical profile based on IETF standards aimed at putting all networks and service providers on the same playing field. The formal adoption by the SIP Forum Board of the SIPconnect Technical Recommendation Version 1.1 is based on recognition that the document has been through comprehensive peer review by known technical experts, including broad membership and significant community review, that it is stable and is well-understood, and that it is believed to have resolved known design choices.

SIPconnect 1.1 contributing companies included Acme Packet, AT&T, Avaya, Bandwidth.com, Boeing, Broadsoft, CableLabs, Cablevision, Cbeyond, Cisco, Columbia University, Comcast Cable, Cox Communications, Digium, Encore Software, GENBAND, Global Crossing, Huawei, Ingate Systems, MetaSwitch, Microsoft, NeuStar, Nokia, Nortel, PAETEC, Panasonic, Pbxnsip, Polycom, Radvision, Samsung, Siemens Enterprise Communications, Sonus Networks, Tekelec, Tele2 Nederland, Voxeo, and XO Communications.

VoIP and SIP Trunking - The Security Issues

VoIP Monitor — Mon, 16 May 2011 16:31:50 GMT

For those of you looking to replace PSTN connectivity, so you can take full advantage of VoIP, giving you much cheaper local, toll-free, domestic and international long-distance services, then SIP trunking can help you save money. That’s why it’s increasingly popular.

SIP trunks eliminate costly time-division multiplexing trunks and gateways, allowing calls to be routed over the carrier’s backbone and use the same IP connection for all communications.

But the caveat is that it needs to be secured. With most VoIP systems, the PSTN serves as a barrier between a company and the outside world, minimising the risk of attack from the Internet. If SIP trunking replaces the PSTN, then that barrier is removed and your phone system becomes vulnerable to IP-based attacks through the SIP trunk.

Security issues around SIP trunking include whether or not you have the same security requirements and security policies as your provider; what changes might have to be made to the firewall, NAT device, IP PBX, private IP addresses, numbering plan and other components; and how you will maintain user/caller ID privacy.

You wouldn’t contemplate connecting your data network to the Internet just relying on the router for security. Everyone has a firewall for good reason. Similarly, protecting your SIP connection is crucial.

To ensure security, you need to deploy a real-time security solution which provides comprehensive threat protection, strict policy enforcement, robust access control, and privacy.

Some data firewall suppliers have now extended their solutions to meet some of the security requirements for connectivity. However, as in all things to do with security, it is sometimes better to deploy specialist solutions for specialised requirements.

Companies like Sipera are now providing solutions which specifically address UC security issues, including SIP trunking. Sipera’s UC-Sec appliance solution, for example, serves as the demarcation point for the client’s VoIP and UC network, enforcing fine-grained security policies.

It protects against SIP and RTP threats, by blocking them at the enterprise perimeter. It maintains the privacy of the internal network, caller/user IDs, and communications, as well as performing firewall/NAT traversal to simplify the deployment of SIP trunks.

Tpad Launches State-of-the-Art VoIP SIP Trunking Solution

VoIP Monitor — Fri, 13 May 2011 17:32:16 GMT

This system is able to combine data, voice and high-speed Internet connectivity over one connection, thereby providing existing and potential customers significant cost-savings while enjoying the benefits of VoIP.

SIP is a service that is deemed more cost-effective than any other telephony solutions available today. It is a streamlined next generation IP telephony solution which allows exchange of voice traffic through Internet connectivity rather than traditional physical cables or PRIs, which translates to cost savings and enhanced efficiency.

Tpad Sales Director Simon Jones explains the benefits of SIP Trunking further by saying, "What Tpad offers with SIP Trunking is not only about saving, it is also about improving workforce mobility and increased productivity as we give each of them connectivity at a fraction of a price. Employees can make phone calls and can be reached through a single number that is accessible at multiple locations so that your customers, co-workers and vendors can get prompt responses."

Tpad's SIP Trunking is offered in various packages that meets the varying needs of small and medium sized enterprises as well as large corporations. Marketing Manager Steven Johns have more to say about their SIP packages, "We aim at providing customers a complete all-in-one solution for their every business telecom need. Our SIP Trunks are offered with synchronous high-speed Internet connection that assures lightning-fast services, not only in the upload and download of data, but in routing calls as well." He goes further explaining, "In fact, our customers will be pleased to know that SIP Trunking isn't only limited to voice communication; it also encompasses sophisticated Instant messaging, toll free number accessibility, media conferencing, and many more highly advanced features."

SIP Trunking by Tpad is backed by Internet connection services offered by its partner company, Supanet, a leading Internet service provider boasting over over a million subscribers to date. Tpad and Supanet has just recently sealed the deal with this partnership to offer customers a fully-equipped, excellent-quality and higly-efficient IP network that allows maximum productivity.

Tahir Mohsan, Tpad SEO says, "Tpad has been very consistent with the development of the most advanced and cost efficient data and communications solutions so that the business sector can operate more efficiently." He concludes, "With our new SIP Trunking service, businesses large and small can save big while enjoying the most technologically advanced services that can be offered by VoIP."

4PSA Brings Carriers New Business Opportunities

VoIP Monitor — Thu, 12 May 2011 15:41:47 GMT

4PSA announces the revamping of its Certified Trunk Provider Program, a partnership program through which SIP Providers can attest the compatibility between their SIP trunking solutions and 4PSA's award-winning VoipNow Unified Communications Platform.

4PSA Certified Trunk Providers will have their services showcased directly in the VoipNow web interface through provider templates. These templates simplify SIP channel configuration, are automatically pushed to all VoipNow installations in real-time and they are easily accessible to VoipNow administrators.

Currently, there are two levels of certification for Trunk Providers: free Silver and advanced Gold. The Gold level of certification includes digitally signed SIP Trunking templates that can be safely distributed over the Internet, as well as the preferential positioning of the templates in the VoipNow interface.

4PSA is continuously searching for SIP providers that are able to deliver the best experience to its customers. Companies interested in applying for this program can find more details here: http://www.4psa.com/company-partners.html.

VoipNow Professional, the winner of 2010 Internet Telephony Product of the Year Award and Unified Communications Magazine's 2010 Product of the Year Award, is a flexible, high-performance solution designed to deliver Unified Communications to companies of all sizes.

ITP VoIP Provides Hosted PBX and SIP Trunking Business Plans

VoIP Monitor — Wed, 04 May 2011 19:00:31 GMT

ITP VoIP is offering two digital phone plans custom-tailored for businesses. The hosted PBX and SIP trunking plans both start as low as $19.99 and allow business owners to choose between managing their systems in-house or letting ITP VoIP handle everything remotely.

Many businesses - small businesses in particular - are not interested in hiring an additional person or persons for an IT department to handle their phone and computer systems, so ITP VoIP offers to do it for them with its VoIP hosted PBX service.

The hosted PBX plans begin with the Metered Plus plan at under $20 per month. From there, clients can upgrade to a selection of Unlimited Plus plans offering support for multiple business lines, as well as unlimited calling through the U.S. and Canada. The hosted PBX packages include conference calling, caller ID, online call log viewing, number portability, music on hold, and individual extensions, among other features.

For businesses who do prefer to manage their phone systems on-site, ITP VoIP offers a variety of SIP trunking plans starting at under $20 a month. The SIP trunking plans also feature a number of add-on services, like an auto attendant, a toll-free number, professionally recorded messages, and paging/remote paging. This is a preferred option for clients with existing phone systems on-site that are interested in saving money on their phone service without switching to hosted PBX.

Both the hosted PBX and SIP trunking plans come with Global Package options: the Basic Package, which offers 1000 free global minutes; the Gold Package, which includes 2500 free global minutes; and the Platinum Package, which provides 5000 free global minutes. These minutes currently apply to 60 countries around the world.

No matter the plan, clients can rest assured their businesses are in good hands. Sporting stellar online reviews, ITP VoIP provides attentive customer service ideal for users unfamiliar with digital phone technology. Dedicated as always to making sure its clientele is happy, ITP VoIP works with its customers to keep them as involved - or uninvolved - as they wish to be.

2011 Touted the Best Year for SIP Trunking Says Tpad

VoIP Monitor — Wed, 04 May 2011 18:58:03 GMT

Tpad announced its confidence that SIP trunking services will reach its all-time high demand in 2011. It announced that with companies all over the world seeking to improve existing telephone systems, it is only a matter of time for SIP trunking to be integrated fully into the telecommunications industry and deployed in business enterprises worldwide. "We are positive that this new service we provide will become widely accepted, especially for companies that make and take calls from clients, local and international," says Tpad CEO Tahir Mohsan.

SIP Trunking is a telecommunications technology that allows convergence of voice and data into one single, merged line. The SIP trunk will serve as a converter that enables the data network to access voice traffic through Internet connectivity. With SIP Trunking, companies can enjoy local and long distance calls, video conference, E911, caller ID, and directory listing features with their current telephony systems without shelling out big investments. Calls are specifically routed through a VoIP network so call charges are significantly reduced. Tpad Marketing Manager Steven Johns adds, "This new technology allows enhanced use of our clients' existing IP or IP-PBX phones without the need for additional physical hardware like PSTN circuits to convert TDM to voice."

Tpad also announced that their SIP Trunking services come in a variety of packaging, offer varying budget options for companies large and small. "We aim to increase customer confidence in SIP technology, so we are offering our services at highly competitive prices to suit existing and potential clients' telephone solution needs," says Simon Jones, Sales Director of Tpad. "In fact, our clients will be pleasantly surprised to know that with just a small amount of investment, we can offer ROI in less than six months from initial deployment," Jones adds.

SIP Trunking is highly recommended for enterprises like BPO or multinational companies with highly-distributed network and a global clientele. SIP trunking can significantly decrease the cost of communications within and outside the business by utilizing better connectivity. As Tpad CEO puts it, "With the increased globalization of businesses worldwide and the trying economy, this technology will be widespread this year."

IPitomy Announces SIP Trunking Interoperability with Windstream

VoIP Monitor — Tue, 03 May 2011 17:26:03 GMT

Windstream and IPitomy announced that they have successfully completed interoperability testing between SIP products and services. Windstream is now interoperable with IPitomy’s Pure IP PBX platform.

IPitomy designs and manufactures IP telephony equipment for businesses. Windstream delivers telecommunications services to residential and business customers in 29 states and the District of Columbia.

IPitomy’s PBX and Windtream services create a complete telecommunication solution for businesses of all sizes using SIP trunking as well as traditional TDM services.

Windstream can now provide new customers using IPitomy’s IP PBX high quality bandwidth and SIP telephone service at an attractive price and unparalleled quality of Service.

Sipera Systems and voipGATE Partner to Secure Enterprise Unified Communications and SIP Trunks

VoIP Monitor — Thu, 28 Apr 2011 15:41:30 GMT

Sipera Systems and voipGATE announces a partnership to better secure enterprise UC and SIP trunks. The partnership enables voipGATE customers to benefit from comprehensive UC security through the Sipera UC-Sec appliances, and the Sipera E-SBC session border controller, the industry's first device for SIP trunk termination to include a unique enterprise VoIP integration module for rapid installation and implementation.

voipGATE's greatest customer advantage is the cost savings from inbound and outbound telephone charges. This translates into immediate savings for businesses. Many voipGATE clients have found cost reductions of 25 percent or substantially higher as a result of using voipGATE as their SIP VoIP provider. Conventional local and international telephone services costs are a sizable percentage of most businesses' current telecom budget. voipGATE SIP Trunking Services are designed to save money while delivering excellent quality.

voipGATE's VoIP solutions delivered with VoIP SIP trunking enable companies with multiple offices anywhere in the world to call free of charge between offices as long as voipGATE VoIP Service is installed at each location. Deployment of the service delivers significant savings to customers that have measurable amounts of existing interoffice communication utilizing traditional telephone company services that include international long distance rates.

By joining Sipera's and voipGATE's competencies, customers will now also be able to secure their communications by using Sipera's groundbreaking solutions, thus having a double business benefit through cost reductions and improved security.

Sipera's UC-Sec appliances proactively address security risks associated with UC applications such as VoIP, instant messaging, IP video and collaboration tools, enabling government agencies and enterprise customers to adopt UC applications and apply comprehensive security. Sipera's groundbreaking "Borderless UC" architecture enables all communications to be encrypted and compliant on any UC device, at any internal and external location.

Tpad and Supanet Join Forces to Offer Premium Business SIP Trunking Services

VoIP Monitor — Wed, 27 Apr 2011 15:54:59 GMT

With the partnership with Internet service provider Supanet, Tpad can now provide business-grade connectivity that will make more efficient use of the an organization's existing PBX Tpad telephone systems. This coming together of sorts represent a marriage of two integral services that the business industry needs most for high quality communication lines within the company and out. As Tpad CEO Tahir Mohsan puts it, "Tpad is committed to providing quality services that aptly meets the needs of the times. We continue to seek towards addressing the needs of our clients and the businesses we cater to, which is why this partnership became possible. We are looking forward to even more innovative solutions in the near future."

Tpad specializes in voice over IP telephony systems which enable an advanced communications model that can serve the needs of large and small companies worldwide. With many years of experience and expertise in the field of telecom, they offer planning, installation and servicing of office telephone systems that are not only cost-effective, but highly adaptable as well. They offer a wide selection of PBX packages that are efficient, highly upgradable and compatible with digital or IP lines. Aside from PBX systems, Tpad also provides a variety of advanced business telephony services ranging from virtual numbers, SIP trunks, and number porting.

On the other hand, Supanet, in operation since 1998, is one of the most highly experienced and coveted Internet service provider in the UK. With over a million registered subscribers, they offer a wide range of Internet connectivity and network options to meet the needs of medium to large businesses. These connectivity options include Dark Fibre, Broadband, Wireless Point to Point Links, Wi-Max, Dedicated or Leased Lines, VPN and DSL. The company prides of its extensive experience in building high-speed networks, installation of network hardware, network planning, as well as consultancy on connectivity needs - from the simplest to the most complex. The company designs and install network systems that are tailored to the needs of the business.

What will businesses expect with combined efforts of these two companies? Traditionally, there seemed to be a void that splits data and voice infrastructure in almost all organizations. This means separate expenditures for hardware involving data, and a different set for voice. However, in the recent years, the walls between data and voice have started to fall, thanks to efforts of companies like Tpad and Supanet. Businesses can now slowly move towards convergence and shared infrastructure for bigger savings and increased functionality. This means that companies can use their data network to access the telephone system and make it work towards increased efficiency and productivity.

One particular communication service that will be made even more cost-effective with this partnership is SIP Trunking. SIP Trunking is one of the most highly-advanced solutions today as it makes better use of data connectivity to allow improved PBX systems. This service offered by Tpad (and now by Supanet as well) allows PBX systems access to VoIP outside the enterprise network through Internet connectivity. Ultimately, it allows a company's VoIP system to talk to the outside world, which in technological terminology is called the PSTN. This methodology eliminates the need for expensive multiple ISDN lines and is thus far less costly, more highly efficient and significantly easier to manage. SIP trunks require fewer hardware pieces and may be purchase in increments, which translates to better cost savings than other alternative solutions. SIP Trunking can also ensure business continuity, geographical openness, flexibility and centralized technology for easier infrastructure management.

SIP Trunking, among other communication solutions, are expected to be made even more efficient when Tpad works exclusively with Supanet. Together, the two companies design a platform that further improves the current capabilities of data infrastructures of clients. "We are happy to announce this partnership, we are sure that as a Team, Tpad and Supanet will propel telecom systems forward," says Steven Johns, Tpad Marketing Manager. He adds, "Products and services that are operated and designed by separate providers tend to lead to compatibility issues, confusion, which then leads to waste of time and money. We have made this partnership possible to ensure more reliable and more efficient telecommunications experience, which is important to the success of the business marketplace."

Through Tpad and Supanet, businesses can make use of voice lines, Internet and SIP services fully integrated with data services, VoIP, managed telephone systems and unified communications internally and externally. These companies have earned high customer satisfaction and loyalty ratings in the country, which make them leading providers in the telecommunications industry.

Companies interested in making their PBX and VoIP systems more efficient through SIP Trunking and similar advanced methodologies can get in touch with either Supanet or Tpad. Faster connectivity and better telecom systems await them after three steps. First, consultant from Supanet or Tpad will assess your existing network and telephone set-up; ask about your business priorities, current needs and future plans, then suggest a package that best suits your business. The assessment will lead to a recommendation which includes a quotation of fees and prices for better decision making. The agreed package is then deployed and installed seamlessly with very minimal business disruptions. Upon complete installation, the services provisioned will be implemented with utmost care along with ongoing technical support.

OnSIP Announces Free SIP / XMPP Accounts

VoIP Monitor — Tue, 26 Apr 2011 16:50:14 GMT

Junction Networks announces a new service, getonsip.com, with the sole mission of providing free SIP addresses to the world. With a getonsip.com address, users can communicate with anyone on the Internet via the open protocol SIP.

The team at Junction Networks hopes to encourage more people to “get on SIP” with getonsip.com. By simply filling out a few fields, visitors can get a SIP address in the form of name@getonsip.com for free. They can then register a SIP device or softphone with the credentials sent to them via email. That’s all users need to start making and receiving free calls to and from friends, family, and colleagues who are also using SIP.

An added benefit to the getonsip.com address is that it’s also a jabber IM address. Users can simply sign into an XMPP client(e.g. Adium, Gajim, Pandion, Pidgin, or Psi) with their getonsip.com address and password.

To acquire a free SIP/XMPP address and learn more about SIP phones, visit getonsip.com.

Mirial ClearSea Brings Standards-based Video Conferencing to Motorola Xoom and HTC ThunderBolt

VoIP Monitor — Wed, 20 Apr 2011 19:49:49 GMT

Mirial announces that Motorola Xoom and HTC ThunderBoltT have been added to the list of certified devices for Mirial ClearSea.

Motorola Xoom is the first Android 3.0 Honeycomb tablet. It features dual core processor and 10.1" widescreen HD display, 2-megapixel front camera and 5-megapixel back.

ClearSea is the only professional video conferencing solution including a software client for Android devices that enables organization to connect Pc, Mac, Android or iOS devices and any standards-based H.323/SIP equipment, such as videoconferencing room systems.

The Xoom and the ThunderBolt are the latest addition to the list of certified devices for ClearSea, which already counts most of the Apple devices, including iPad 2 and several Android smart phones and Tablet such as the Samsung Galaxy Tab, the Motorola Atrix 4G, the Dell Streak and many others.

ClearSea for Android, as well as for iOS, Windows and Mac, is available today.

Cbeyond Chief Technology Officer Chris Gatch Elected New Term on SIP Forum Board of Directors

VoIP Monitor — Mon, 04 Apr 2011 17:30:19 GMT

Cbeyond announced that chief technology officer Chris Gatch has been re-elected to the Session Initiation Protocol Forum Board of Directors. The SIP Forum is an industry association composed of decision makers from leading IP companies that seeks to further the interoperability and adoption of products and services using SIP communication tools.

During previous leadership roles with the SIP Forum, Gatch engineered the transition of the SIPconnect Interface Specification, which facilitates connections between users and service providers, from Cbeyond to the SIP Forum. In 2006, a team led by Gatch released the SIPconnect Technical Recommendation establishing the first industry standard for SIP trunking between IP PBX and service provider networks. During his new term he looks forward to supporting the SIP Forum’s roll out of a major update to SIPconnect in version 1.1.

“A good sign a technology standard is gaining acceptance is when it evolves to the point of helping vendors, providers and customers resolve their biggest issues,” said Gatch. “SIP has become the industry standard which enables small businesses to control their real-time multimedia communications without sacrificing quality or money. I will continue to work with the SIP Forum to uncover additional ways the technology can meet the needs of small businesses.”

A veteran of the telecommunications industry with more than a decade of VoIP technology experience, Gatch is serving his third term on the SIP Forum Board of Directors.

Cbeyond was the first VoIP service provider to embrace SIPconnect and was also one of the first to launch successful SIP trunking capabilities. Today, Cbeyond offers a full suite of BeyondVoice services with SIPconnect to provide the IP phone and data services necessary for efficient business operations.

VoIP Services Market Nears $50 Billion Mark; Breakout Year for SIP Trunking

VoIP Monitor — Fri, 01 Apr 2011 19:36:44 GMT

Market research firm Infonetics Research this week released VoIP and UC Services and Subscribers, a market share and forecast report that includes two Business VoIP Service Provider Scorecards that will be published later this year, and an IP Centrex Provider Tracker highlighting deployments by provider, region, service, and platform.

ANALYST NOTE

“The VoIP service market weathered the economic turmoil of the last couple of years, and, with increasing customer adoption, reached $49.8 billion in 2010 (compared to $34.8 billion in 2008). While the residential services segment remains the largest of the market at 69% of total revenue, business VoIP services are growing at faster rates; a notable example: SIP trunking had a breakout year with 143% revenue growth in 2010,” notes Diane Myers, directing analyst for VoIP and IMS at Infonetics Research.

VOIP SERVICES MARKET HIGHLIGHTS

Infonetics Research forecasts the combined business and residential/SOHO VoIP services market to grow to $74.5 billion in 2015
Managed IP PBX business VoIP service revenue is expected to more than double from 2010 to 2015
NTT of Japan retains its leadership as the world's largest residential VoIP service provider, followed by Comcast and France Télécom
The fastest growing segments of the VoIP services market are SIP trunking and hosted UC telephony
The number of residential VoIP subscribers increased 19% in 2010 to 157 million worldwide
Based on healthy demand for cloud-based services, the number of seats for IP Centrex and hosted UC services grew 20% in 2010

IP Telephony: Deploying VoIP Protocols and IMS Infrastructure - Now Updated with Coverage on SIP and the IMS Infrastructure

VoIP Monitor — Wed, 16 Mar 2011 17:46:11 GMT

Research and Markets has announced the addition of John Wiley and Sons Ltd's new book "IP Telephony: Deploying VoIP Protocols and IMS Infrastructure, 2nd Edition" to their offering.

All you need to know about deploying VoIP protocols in one comprehensive and highly practical reference - Now updated with coverage on SIP and the IMS infrastructure

This book provides a comprehensive and practical overview of the technology behind Internet Telephony, providing essential information to Network Engineers, Designers, and Managers who need to understand the protocols. Furthermore, the author explores the issues involved in the migration of existing telephony infrastructure to an IP - based real time communication service. Assuming a working knowledge of IP and networking, it addresses the technical aspects of real-time applications over IP. Drawing on his extensive research and practical development experience in VoIP from its earliest stages, the author provides an accessible reference to all the relevant standards and cutting-edge techniques in a single resource.

Key Features:

Updated with a chapter on SIP and the IMS infrastructure
Covers ALL the major VoIP protocols SIP, H323 and MGCP
Includes a large section on practical deployment issues gleaned from the authors own experience
Chapter on the rationale for IP telephony and description of the technical and business drivers for transitioning to all IP networks

This book will be a valuable guide for professional network engineers, designers and managers, decision makers and project managers overseeing VoIP implementations, market analysts, and consultants. Advanced undergraduate and graduate students undertaking data/voice/multimedia communications courses will also find this book of interest.

SIP/H.323 Video Conferencing on iPad 2 with Mirial ClearSea

VoIP Monitor — Mon, 14 Mar 2011 17:50:23 GMT

Mirial is glad to announce that Mirial ClearSea can be seamlessly used also on iPad 2, taking full advantage of the new front facing camera with the possibility of switching between cameras even during the video call.

ClearSea is the first professional video conferencing solution providing a software client for iOS devices, enabling organization to connect any Pc, Mac, Android or iOS device and any standards-based H.323/SIP equipment. It allows to integrate traditional room based video communication systems or to deploy a new HD desktop and mobile solution without the need to invest in expensive infrastructure.

"We have been waiting for the addition of a front facing camera on the iPadT" said Cristoforo Mione, VP Marketing at Mirial. "Now that the iPadT is ready for true video conferencing, our software transforms it in a standards-based video endpoint."

ClearSea for iOS devices, including iPad 2, is available today.

GENBAND Solution for SIP Trunking Achieves Microsoft Office Communications Server Qualification

VoIP Monitor — Mon, 14 Mar 2011 17:40:11 GMT

GENBAND announces that the SIP Business Trunking softswitch solution offered on its market-leading GENBAND C20 Converged Softswitch has achieved interoperability qualification with Microsoft Office Communications Server 2007 R2. Additionally, GENBAND announced that its C20 softswitch is an integral component of the Swisscom SIP Trunking Solution which was recently certified for use with the Microsoft Lync Server 2010. Swisscom, Switzerland’s leading telecoms provider, announced last month that Microsoft certification of its VoIP Gate service enables the Company to offer its corporate customers a complete IP-based communications solution that is both innovative and reliable.

Microsoft Unified Communications Open Interoperability Program ensures that customers have seamless experiences with setup, support and use of qualified telephony infrastructure and services with Microsoft Corp.’s unified communications software. With more than 650,000 ports deployed globally, GENBAND’s solutions for SIP trunking have been tested with the majority of PBX solutions available in the business market.

GENBAND’s solution for SIP Business Trunking allows service providers to deliver flexible and low-cost IP-based communications between an enterprise PBX and the public network. With the GENBAND solution, service providers are able to support multiple voice sessions that are managed under a single account for billing, maintenance and service assurance. In addition, by eliminating the need for separate voice and data connections, SIP Business Trunking helps reduce costs for enterprise users and can lay the groundwork to offer subscribers a variety of real-time applications including Unified Communications, Instant Messaging, Network Presence, Mobility, Messaging and Desktop Integration.

Versadial Adds SIP Trunk Integration to VS Logger Call Recording System

VoIP Monitor — Fri, 11 Mar 2011 16:53:18 GMT

Versadial Solutions announce the release of SIP trunk capabilities for its VS Logger call recording system. Call data can now be captured directly from the SIP trunk, including start/stop times, caller ID or dialed number and other details.

The Versadial recording solutions are designed to help companies in a variety of industries, including public safety and 911 emergency response, financial services, contact centers, health care, legal services, and all types of small and medium sized businesses. Compatible with business communications system providers, including Avaya, Cisco, NEC, Alcatel, Nortel, Panasonic, Siemens, Mitel, ShoreTel and Toshiba. VS Logger and Adutante applications can help companies to improve risk mitigation, quality assurance, customer retention, dispute resolution, regulatory compliance requirements and other critical business concerns.

IntelePeer SIP Trunking Earns Qualification to Support Microsoft Lync Unified Communications

VoIP Monitor — Wed, 09 Mar 2011 18:11:48 GMT

IntelePeer announces its SIP Trunking services have been qualified for interoperability with Microsoft Lync 2010. Formerly known as Office Communications Server 2007 Release 2, Lync is the next generation of Microsoft's Unified Communications services with feature-rich multimodal communications capabilities.

The qualification allows IntelePeer, a Microsoft partner, to provide a turnkey SIP Trunking solution for enterprises using Microsoft Lync by delivering a seamless connection to any Lync or OCS deployment federated via IntelePeer's global SuperRegistry directory as well as to any Public Switched Telephone Network communications device reached through IntelePeer's Peering Grid services. IntelePeer's services, available through the company's channel partners, are interoperable with any Lync deployment and are consistent and supportable in accordance with Microsoft's rigorous standards.

IntelePeer provides a comprehensive suite of on-demand cloud communications including SIP Trunking services that are backed by deep IP communications expertise. Through its cloud-based platform, IntelePeer delivers feature-rich, carrier-quality services over public or private Internet Protocol connections using the scale of the cloud. The company's SuperRegistry directory of more than 478 million phone numbers enables intelligent communications Peering Grid services that directly transport billions of voice minutes every month and enables the delivery of rich multimodal communications, such as video calls between peered wireless, wireline and VoIP service providers, contact centers and enterprises. IntelePeer's patented cloud platform provides real-time service availability, eliminating the lengthy delays associated with typical service deployments.