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.

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.

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 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.

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%.

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.”

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 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.

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%.

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.

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 everythi

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.

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.

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.

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.