A recent study said that while SIP trunking has seen steady adoption over the past five years, interoperability issues and a lack of standardization have limited its overall adoption and hamstrung service providers from gaining deeper market penetration.

But a new technical initiative from the SIP Forum--SIPconnect-IT--aims to speed up vendor and service provider interoperability with the SIPconnect 1.1 Technical Specification, ratified by the SIP Forum earlier this year.

A series of testing events will help provide application developers, equipment vendors and operators with a platform from which to demonstrate and test implementations of the specification in real-world scenarios.

The first SIPconnect-IT testing event is planned for mid-2012 in Louisville, Colo., and will be hosted by CableLabs.

The ratification of the SIPconnect 1.1 technical recommendation earlier this year was seen as a significant step toward the adoption of a common interoperability specification between SIP-enabled IP-PBXs and service provider networks. It provides a 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 best-practices guidelines for vendors and service providers as they develop new equipment and IP applications for deployment.

"SIPconnect-IT represents the next step and is designed to transform these specs into action and to ensure IP applications and infrastructure utilizing the SIPconnect 1.1 guidelines can be implemented in live telecom environments," said Marc Robins, SIP Forum president and managing director.

For more:
- see this release

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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 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 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, known for its VoipNow Unified Communications platform, has announced the public availability of Cloud Telephony, the flexible, next-generation SIP trunking service that "can be provisioned within minutes."

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.

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.

The benefit of open standards is that it put the customers in control. They can use any SIP-compatible phone with any SIP-compatible server. We can place calls between companies. We can freely choose a SIP PSTN-gateway service provider. There are no vendor lock-ins. To reach this, a lot of work is needed. A lot of [...]

Level 3 Communications has announced that it is working with Microsoft to provide SIP Trunking with nomadic E-911 (enhanced 911) designed to integrate with Microsoft Lync.

Junction Networks has announced that MBLM NYC has become the company's 10,000th customer of its hosted VoIP services. Following a merger, the New York-based branding firm chose to deploy OnSIP Hosted PBX service to "quickly deploy a complete communications solution and save on upfront investment."

snom, a German developer and manufacturer of SIP devices, has announced the release of snom ONE - a new family of IP PBX systems that runs on standard servers. According to the company, snom ONE will provide customers with an IP telephony system which supports all the functionalities of snom IP phones.

IPsmarx Technology and VoIP Innovations have announced compatibility and API Integration for local and toll free DIDs. According to the companies, this integration will enable VoIP service providers to easily and quickly offer services to a new market.

Two weeks after releasing fringOut that enables users to make cheap calls to any regular landline or mobile phone anywhere in the world, with rates as low as 1¢ a minute, fring announced the service is now available for Android phones.

Avaya and Skype have announced a strategic agreement to deliver communications and collaboration solutions to businesses of all sizes. The multi-phase deal includes both go-to-market and a joint technology integration.

Grandstream has extended the portfolio of its GXP series enterprise SIP telephones with the introduction of the new GXP2110. Based on Grandstream’s broadly interoperable SIP stack, “the GXP2110 SIP telephone delivers superior HD audio quality for crystal clear voice communications, packed IP telephony features and integrated Web applications, as well as support for highly flexible XML customization and strong security protection,” as the company says.

Grandstream Networks has extended the portfolio of its popular GXP series enterprise SIP telephones with the introduction of the new GXP2110. Based on Grandstream’s broadly interoperable SIP stack, the GXP2110 SIP telephone delivers superior HD audio quality for crystal clear voice communications, packed IP telephony features and integrated Web applications, as well as support for highly flexible XML customization and strong security protection.

Extending Grandstream’s commitment to quality, feature, and performance, the new GXP2110 SIP phone comes standard with HD handset and high performance full duplex speakerphone, a broad range of voice codecs, dual network ports with integrated PoE, 4 line keys, 3 soft keys, 18 programmable BLF keys, 5-way conferencing, large 240x120 backlit graphical LCD with 8-level grayscale, multiple languages, large phone book and call log (2,000 records), automated provisioning using TR-069 and encrypted XML file, extension module expandability, and a number of integrated Web applications such as real-time local weather, stock, currency, RSS news, etc. In the near future, more advanced Web applications will continue to be integrated via FREE firmware upgrade and an open Web service API will also be provided for advanced custom enterprise/Web application development. Grandstream plans to continue to expand the GXP phone series with new models during the rest of this year.

Pricing and Availability

The GXP2110 is commercially available for purchase now through Grandstream’s worldwide distribution channels at a MSRP of US$139.

VoicePulse will be sponsoring FusionPBX’s first official training sessions by providing SIP Trunking services to its participants. The three day seminar will be led by Mark J. Crane, an experienced technical trainer and creator of FusionPBX. Participants will receive in depth training in a variety of topics including setting up extensions, call routing, IVR menu, dialplan, security and more. Basic and advanced setup of a SIP provider, such as VoicePulse, will also be covered.

SIP Trunking is a virtual phone line that utilizes a broadband connection and SIP to deliver inexpensive local, toll-free, domestic and international long distance service through an IP network. VoicePulse SIP Trunking is a flexible and reliable SIP solution that works with virtually any SIP based device such as an IP PBX, softphone or mobile SIP application. In addition to SIP Trunking solutions for the small-medium enterprise customer, VoicePulse offers carrier services and various reseller programs.

Registration for the training sessions is now open. Anyone interested may register by visiting www.FusionPBX.com. Limited space is available. Price per seat is $1295 and the first 10 people to sign up will receive a discounted price of $995.

fring has just introduced fringOut, a new service that enables users to make cheap calls to any regular landline or mobile phone anywhere in the world, with rates as low as 1¢ a minute. Rates for Canada and UK are even lower -- 0,6¢ and 0,7¢, respectively, US calls starts at 1,2¢ per minute.

Skype on Monday announced the official launch of Skype Connect 1.0, formerly known as Skype for SIP. Previously available in beta, Skype Connect delivers a business solution that enables IP-enabled private branch exchange (PBX) or Unified Communications systems to connect to Skype.

Acrobits recently made a big splash in the VoIP world with their new business caliber SIP client for the iPhone, Groundwire. While this has legions of VoIP users on the iPhone very excited, it was also important to Acrobits to reward users of their already excellent SIP client Acrobits Softphone with an inexpensive upgrade option. Unfortunately, iTunes doesn’t offer a viable option for Acrobits to do this. It looked like existing users of Acrobits Softphone would need to purchase Groundwire at full price to benefit from the exciting new features of Groundwire. Not to mention the licensing agreement for the G.729 codec would require users to purchase a new license to use the codec in Groundwire (G.729 licensing requires a separate license for each client or device).

This was unacceptable to Acrobits. “We value our customers and want to reward loyal users of Acrobits Softphone,” says Acrobits. So for two days only (August 30th and 31st, 2010), Groundwire is on sale for only $2.99. And users who want the benefit of using the low bit rate, high quality G.729 codec will be able to purchase the license at half price, only $4.99. Not only is Acrobits leading the market in mobile SIP applications, but they are proving themselves adept at building and maintaining customer loyalty as well.

Though the promotion is targeted toward their existing customer base, new customers will benefit from the promotion as well. So if you were considering purchasing a SIP client for your iPhone, or were waiting for the right time to delve into the world of VoIP, now is the time to do it. You won’t be disappointed.

Acrobits will continue to bring new and exciting SIP clients to the mobile VoIP user in the coming months. Their Acrobits Softphone client for Android should be available on the Android Market sometime in September and you can expect an iPad specific version by the end of the year. Keep an eye out for this up and comer in the VoIP world.

AT&T announced that new and existing virtual private network (VPN) customers may add VoIP service to the network solution delivered over AT&T's global network cloud. This converged solution is said to enable customers to consolidate their separate voice and data networks, reduce equipment and maintenance costs, and simplify migrating these complimentary capabilities to a common, secure infrastructure.

Acrobits has released their new business caliber SIP Client, Groundwire for the iPhone. It supports transfer and attended transfer, call waiting and conference calling. It also adds a voicemail waiting indicator and a programmable voicemail dialer.

Mobile software developer Acrobits released their new business caliber SIP Client, Groundwire for the iPhone, this week. This news should delight both iPhone owners and the VoIP world in general. SIP users finally have a mobile client capable of meeting all their needs.

Groundwire has all the great features of Acrobits’ current softphone client, Acrobits Softphone, and adds some very important business caliber features. Groundwire supports transfer and attended transfer, call waiting and conference calling. It also adds a voicemail waiting indicator and a programmable voicemail dialer. But that’s just a taste, here’s the full list of features.

• Multitasking background support for iOS4
• Transfer and attended transfer
• Push Notifications, a reliable way to receive calls when the Softphone is closed
• Multi line
• Call waiting, switch between calls with headset controls or just shake the iPhone
• Conference calling
• voicemail waiting indicator with programmable dialer
• Customizable ringtones, choose from our selection or pick something from your media library
• HD Wideband audio through G.722
• excellent sound quality, includes the G.711, GSM and iLBC audio codecs. Make an in app purchase to add G.729 Annex A for great quality over 3G networks
• Completely redesigned audio optimized for the best VoIP experience on the iPhone
• Number rewriting, enabling you to utilize your existing contacts without having to create new entries to satisfy the dialing requirements of your PBX or SIP provider
• Address Book Matching to automatically format your contacts into the proper international format
• TLS support for encrypted SIP
• Bluetooth support for iPhones with OS 3.1 and higher
• audio codec manipulation, enabling you to prioritize the codecs used and disable ones you don’t want to use
• call recorder and player, seamlessly integrated into the call history
• comfortable and super-smooth user interface, fine-tuned specially for the iPhone
• customizable background and colors
• full localization (currently English, Chinese, Danish, Korean, Norwegian, Polish, Portueguese, Swedish, French, German, Italian, Russian, Spanish, Czech and Slovak)
• very easy configuration
• simultaneous registration of multiple SIP accounts, have multiple accounts registered to receive incoming calls and switch the account used for outgoing calls without leaving the keypad
• iPhone contacts integration. Easy to call anyone in your contacts via SIP
• Contacts search function, search your contacts by name or number
• add new contacts directly from the softphone
• Quickdial, your 12 favorite contacts are one touch away
• ability to generate DTMF tones while in call, to control various PBX features or automated systems (use audio, rfc 2833 or SIP INFO)
• speakerphone support
• detailed call history, with intelligent call grouping for an easy overview
• support for sip:username URLs in phonebook
• configurable RTP port range
• SIP Proxy support, VPN support
• STUN server support, automatic service discovery using DNS SRV queries
• quick import of accounts from major VoIP Providers, like Gizmo5, Voipcheap, TerraSIP and others
• phone number resolution. We present the phone numbers in a convenient format with grouped digits, display the flag and country name and even region/city name for some countries, including the U.S.

Acrobits has once again proven their dedication to providing the best VoIP experience on the iPhone. And with their upcoming release of Acrobits Softphone for Android, SIP users on Android will soon be singing Acrobits’ praises as loudly as iPhone users do now.

LG-Ericsson USA officially launched its brand into the North American market offering a broad portfolio of end-to-end data and voice networking solutions for businesses ranging in size from small- and medium-sized businesses to large enterprises.