1 SIP Report Buyer s & Guide Winter/Spring 2011 By David Byrd, Broadvox Byrd Practicing safe SIP Decision-makers are attracted to appealing benefits of Internet telephony such as VoIP/SIP Trunking and Unified Communications as a Service (UCaaS). Advantages include increased productivity, QoS assurance and virtual PBX hosting. However, before proceeding C-level managers often ask if IP communication services are safe. Of main concern is the interconnectedness of the web, resulting in vulnerabilities that could expose networks, communications and databases. Although increased exposure is a consequence of utilizing the Internet, IP voice communications can be integrated with confidence with implementation of safe SIP practices. SIP Forum Page 2 See BYRD, page 3 Ford An interview with LegacySIP.com By Carl Ford SIP Trunking is coming of age in very unusual ways SIP Trunking is in some ways an oxymoron. The real story regarding the combination of SIP with the word trunk, is that the access method has been bundled with VoIP SIP solutions. The reason this has happened is that voice services more often cannot be supported by the data network. Having attended ITEXPO for the last five years I have been in the mode of watching the discussion of SIP Trunking and realizing the gap between the carriers, customers, vendors and resellers. Here is what I see. The concept of SIP Trunking has been embraced by the technologists. Carriers and some ITSPs have been SIP Guide Page 6 very successful in selling wholesale SIP Trunking to companies like Google and large enterprises. Vendors have been integrating SIP Trunking into their solutions and for the customer that is upgrading these systems are widely available. Resellers have been very interested in SIP Trunking, but the reality is that often the price point can be replicated with legacy offers if you include manufacturers costs of PBX modifications and licensing, so the drive to adopt the new solutions is diminished. Customers have been in some ways the last to hear about the opportunity. And the reason is for many the migration to enable SIP trunking is expensive. % Read more from Carl Ford online at and visit legacy.sip.com Horak on SIP Page 2 Safe SIP Safe SIP trunks By Adam Boone, Vice President of Marketing and Product Management, Sipera Systems In enterprise communications the spotlight is on SIP trunks, low-cost communications links leading the evolution to VoIP and UC. Firms can replace more costly TDM-based trunks and extend communications with greater flexibility. VARs are enabling companies in a variety of industries to adopt SIP trunks and migrate to next-generation communications quickly and easily. However, carefully planned security is an essential requirement for reducing exposure to risks like toll fraud and unauthorized network access. In 2009 and 2010 there was a sharp rise in toll fraud, costing tens of millions of dollars. A number of Internet hacking attacks are targeting VoIP systems and looking for improperlysecured SIP trunks. IT trade journals and See SAFE SIP, page 3 SIP to Legacy PBX over existing T1 / PRI connections! Now you can easily deploy SIP at legacy PBX sites Nortel, Lucent, Avaya, Siemens, Tadiran... Whatever you have. It just doesn t matter! Ascend to SIP SIP challenges No Problem We have been at this since the beginning Our low rates plus our on premise device will save up to 70% of monthly T1/PRI costs Multi-site Ÿ Old customers installed years ago Ÿ Customers with wire upgrade concerns Ÿ Cut costs, not quality - Toll quality business-class calls Ÿ Our LSA Trunking Gateway was designed from the ground up to be Plug-N-Play Ÿ 10 minute installs most common - Installation support included Ÿ Our service is remotely monitored and managed Ÿ Remote configuration changes easily made as needed Ÿ Enterprise Softphone support for Presence, IM and Voice/Video communications Let us help you save money! Contact us now for a free, no obligation consultation (877) Ask about our Reseller Program
2 Winter/Spring Telecom Reseller: SIP Report SIPconnect: Providing best practices for SIP trunking By Marc Robins, Managing Director, SIP Forum SIPconnect is a high-profile initiative of the SIP Forum that has contributed greatly towards moving the SIP standard forward. The development of SIPconnect began in 2004, driven by a handful of companies seeking to develop a consistent set of interoperability guidelines between IP PBXs and service providers. In 2006 these companies, including Avaya, BroadSoft, Cbeyond, Cisco, Mitel and Talkswitch, contributed their work to the SIP Forum, which worked to broaden industry support. As a result, SIPconnect now enjoys widespread industry acceptance. The underlying basis of the SIPconnect Technical Recommendation, Version 1.0, ratified by the SIP Forum in January, 2008, is a set of SIP trunking best practices and guidelines to ensure seamless interoperability among participating vendors and service providers. The key driver of SIPconnect is to foster the end-to-end IP connectivity needed for SIP to deliver on its full potential. While telecom vendors are well along the path to IP, many networks remain TDM-based, and by extension this limits the value enterprises can derive from their IP PBXs. While the installed base of telephony systems remains largely TDM-based, the majority of new systems and line shipments are IP-based, and that trend is accelerating. Unfortunately, some service providers have not kept pace, making it difficult for enterprises to fully capitalize on IP s inherent advantages. Before outlining how SIPconnect addresses this issue, consider the value proposition of end-to-end IP. In a full-on IP scenario, enterprises can access the full range of telephony features that an IP PBX can support. In addition to providing feature parity for a TDM PBX, end-toend IP supports value-added services not available on a TDM network. This starts with advanced telephony features, but more importantly enables the integration of telephony with other communication modes, such as , chat and video. SIP has become the protocol of choice for real-time multimedia communication, the value of which is self-evident when compared to how relatively little enterprises can do in a TDM-based environment. Building on this, end-to-end IP allows enterprises to leverage the power of Webbased communications tools such as presence and click-to-talk for a richer experience that goes well beyond what any IP PBX can provide on its own. There is also an important financial aspect to this value proposition. The cost savings on telecom services are wellknown, such as lowering long distance charges, eliminating moves, adds and changes (MAC), and eliminating charges for common features. Beyond that, however, are savings in trunking, as IP networks are more efficient than TDM and do not require physical trunks for every location. Another important financial benefit is that costly media gateways are not required for end-to-end IP, thereby reducing the total cost of ownership (TCO). Finally, SIPconnect yields one standard that all manufacturers and service providers can use and that simply makes it easier to do business. There is no need for interoperability testing with every PBX vendor or every carrier network. This translates into a faster time to market, simpler and faster deployment and a richer set of SIP-based services. While TDM networks will live on for some time to come, there is a solution in the market today that enables end-to-end IP. The transition to SIP Trunking is gaining serious momentum as enterprises look for new ways to economize, which is a key benefit of SIP Trunking. Enterprises can choose to route some or all voice traffic over SIP trunks, and will save money by reducing the number of PRIs used for connecting to the PSTN. SIP Trunking also sets the stage for all the other benefits outlined in this article. The SIP Forum s SIPconnect initiative plays an important role in fostering the adoption of SIP Trunking. In addition to the Technical Recommendation that delivers best practices, the SIP Forum also offers the SIPconnect Compliant Certification Program. This provides participating equipment vendors and service providers with a credible seal of approval, a mark that demonstrates interoperability compliance with an industrybased body. For the first time, both service providers and telecom equipment vendors have a standards-based approach that ensures interconnectivity and interoperability. This program has inherent value on several fronts, but the SIP Forum s work is far from done. Following the ratification of Version 1.0 of SIPconnect, work quickly began on Version 1.1. SIPconnect will continue to focus on supporting voice services and key enhancements for voice in Version 1.1 including improved security, clarification around user identity, and defining registration scenarios. Of course, SIP is a multimedia protocol, and can just as readily support Fax, Unified Communications (UC), video, instant messaging (IM), chat and other real-time applications. However, voice remains a primary mode of communication in the marketplace. As we get closer to an all-ip world, SIPconnect will have a firmer foundation upon which to add these other modes. Support for Version 1.1 is more broadly based than Version 1.0, and reflects how important interoperability is and how it has extended beyond the traditional telecom community. Huawei s Spencer Dawkins, well-known in the SIP community for having authored SIP-related IETF RFCs, has been brought on to edit the Version 1.1 specification. Key corporate content contributors to this updated specification include Avaya, BroadSoft, Cbeyond, CableLabs, Huawei, Microsoft, MetaSwitch and Siemens Enterprise. Importantly, SIP Forum full members Microsoft and CableLabs have contributed as well. As Microsoft has a very strong presence in the enterprise and cable operators see the business market as a significant growth opportunity, these two members significantly broaden the ecosystem for SIP-based interoperability, and will accelerate industry participation in the SIPconnect initiative. Finally, a large number of participants have been very active in the SIP Forum Technical Working Group, submitting comments, suggested edits and other useful information as the work progresses. SIPconnect 1.1 discussion participant companies include AT&T, Cbeyond, Global Crossing, PAETEC, XO Communications, Acme Packet, Avaya, Bandwidth.com, Boeing, Broadsoft, Cisco, CableLabs, Columbia University, Digium, Encore Software, Huawei, Ingate Systems, MetaSwitch, Microsoft, NeuStar, Nokia, Nortel, Panasonic, Pbxnsip, Polycom, Radvision, Samsung, Siemens Enterprise, Sonus Networks, Tekelec and Voxeo. All told, SIPconnect is poised to play an important role in advancing the migration from TDM to IP, and will open up opportunities for a wider range of options from which to choose. Once the base for seamless end-to-end IP telephony has been established, other modes will quickly follow, creating a richer, more cost-effective communications environment that enterprises will no doubt welcome. With this degree of interoperability in place, it will become increasingly difficult for enterprises to justify TDM inertia. % Marc Robins serves as President of SIP Forum LLC (the operating U.S. subsidiary of the SIP Forum) and as the consulting Managing Director of the SIP Forum. Marc has been involved in the telecommunications industry as a reporter and analyst, editor and author, trade show producer and publisher, and marketing executive and consultant 25+ years. With a perspective honed by decades of consistent industry analysis and coverage, Marc is an internationally recognized authority in the field of IP telephony and new IP communications technologies and their commercial applications. Horak SIP Forum: All about SIPlification By Ray Horak, Technology Editor There can be little doubt that telecommunications, in general, is transitioning to the Internet Protocol (IP) and that Voice over IP (VoIP), in specific, eventually will overwhelm analog and TDM protocols and technologies at many levels. A substantial number of enterprise and SMB users have already made the transition. Many, if not most, of the rest will follow in the next decade, if for no other reason than their legacy PBXs and key systems simply begin to fail. Some, of course, need a little encouragement. Metcalfe s Laws states that the value of a telecommunications network is proportional to the square of the number of users (n2) of the system. Robert Metcalfe developed the law to describe the value of Ethernet, but the same logic applies to just about any network and application. So it is that the considerable advantages of an IP PBX are magnified when it interconnects to other IP PBXs, and particularly if the systems interconnect on a peer-to-peer basis and interoperate seamlessly. Thereby, all of the features can, at least theoretically, be made available transparently to all users. To accomplish this minor miracle, the systems and interconnecting WANs must implement a common enabling protocol. The IETF defined the Session Initiation Protocol (SIP) in 1999 as an Application Layer signaling protocol for establishing, modifying and terminating multimedia sessions or calls over an IP network. SIP offers considerable advantages over the ITU H.323 protocol, which is criticized for being overly complex and highly centralized. Over the years, however, SIP has grown to approximately 150 RFCs, many of which include multiple options that can be implemented in various ways. So, SIP has assumed a complexity of its own. The SIP Forum, established in 2000 to promote SIP, focuses intently on resolving issues of interoperability. I recently had the pleasure of interviewing Marc Robins, Managing Director of the SIP Forum, who explained that SIP trunking offers the promise of seamless VoIP in support of presence, mobility, video and other rich media, end-to-end. SIP trunks bypass the PSTN and, therefore, eliminate the gateways that perform conversions from IP to TDM to analog and back again. That certainly eliminates the potential for conversion errors. More importantly SIP trunking maximizes the advantages of the interconnected IP PBXs, avoiding the loss of functionality associated with networking at the level of the lowest common denominator, which in this case happens to be the PSTN. The SIP Forum is a non-profit organization with a growing membership. The nearly 50 dues-paying full members include equipment vendors, cable operators, ITSPs and other service providers. There also are more than 9000 individual members and seven prestigious members in the academic institution category. The SIP Forum works closely with formal standards bodies such as the IETF to influence the evolution of SIP and related standards. The SIP Forum developed the SIPconnect Technical Recommendation and promotes an associated interoperability compliance program. The SIP Forum also sponsors frequent SIPit interoperability test events, and has active task groups focused on Fax over IP interoperability, user agent configuration, and SIP in the Smart Grid. % For more visit CALL ACCOUNTING
3 Telecom Reseller: SIP Report 3 SAFE SIP BYRD Continued from page 1 Continued from page 1 regional media sources are also routinely reporting small companies receiving toll fraud-inflated phone bills. Resellers have an opportunity to help customers avoid risks with these services and products: Proactive SIP security planning: deployment is increasingly widespread and a set of well-understood application-layer security best practices have emerged. Session border control: a SIP trunk should be terminated on-premises by a device that includes session border control functionality to control the demarcation points between networks. Application-layer threat mitigation: ability to track and monitor trunk utilization and user behavior and identify patterns that indicate toll fraud attacks. A signature-based SIP security device can block unauthorized access, Vishing, Spam-over-Internet Telephony (SPIT) and other common security threats. SIP vulnerability assessments: periodically audit the SIP security architecture and modify it as necessary. % The transition from TDM to IP exposes weak infrastructures. There are more endpoints which attract interest from crackers. IT departments and VARs, need to make every effort to prevent or minimize disruption or piracy by evaluating security environments, settings and access codes. Dan York, in his book Seven Deadliest Unified Communications Attacks, describes those a company could be susceptible to if unprepared. He explains how UC, which consists of IP-based equipment, applications and integration of voice, video and data, is exposed to the same security issues as data. Therefore managers must be ever vigilant and apply similar procedures for insecure endpoints as those implemented for access to business networks and applications. York notes that despite such threats from outside, ones from within a company are most prevalent, costly and difficult to prevent. Man-in-the-middle style attacks are a larger threat than all others, because the easiest place from which to eavesdrop or alter information is from within the enterprise itself. The sophistication required to access voice and data packets is considerably more expensive and difficult than the ability of employees and others to eavesdrop, pilfer, modify or appropriate confidential data. So why don t we read more about employees stealing company secrets and selling them to competitors, or giving documents to a colleague in order to help them beat the system? Because incidents of inter- For more visit sipera.com. Bradshaw Page 17 UC Networks More at Winter/Spring 2011 eseller. telecomr/sip com Products nal abuse are underreported, in an effort to avoid the costly repercussions of a loss of trust in management. Internal breaches occur, and the cost of lost business continued to be the most expensive consequence of a security breach in In order to avoid the loss of trust and reputation, it is important for IT departments to apply the same standards for voice security as they do for data. Internet telephony services can be safeguarded to ensure security and limited access. End users can achieve safe interoperability between VoIP/SIP Trunking services and their individual networks. The first step in practicing safe SIP involves managing administrative access and insisting upon use of strong passwords to address unintended access or actions. Additional steps include assuring laptops and servers are properly configured, backing up databases with strong encryption, and insisting on regular password changes. Maintaining patch updates when weaknesses and viruses are discovered is essential. Supplementary or managed security solutions and Session Border Controllers with VoIP security features are available. Securing broadband connectivity through your SIP Trunking provider minimizes exposure to the public Internet and maximizes QoS. All this can help the coupling of IP telephony services with business networks to be safe from breaches, attacks and viruses. % iscoord UC client iscoord, provider of Unified Communications clients, announced is-phone for iphone/ipad/ipod Touch v2.0 that interoperates with IP-PBXs and IMS platforms. Markus Sieger, CEO, said, Many companies are building mobile communication on Apple devices and we are proud to integrate an enterprise-class UC feature set on all ios4 devices. Carriers, large enterprises and other service providers can now offer a complete family of UC applications on one foundation to their clients on all platforms. Our new app was developed with the help of customers and large telecom partners to fit corporate needs, said Rolf Kuster, Head of Development. Employees like to use their mobiles with the same feature set as on their deskphone and need to connect to their IP-PBX, softswitch or IMS platform. is-phone is closing this gap with a software-based approach, enabling enterprise-class IP telephony for companies of all sizes. Features include: Multiple SIP account and call support, conference calls with up to 4 participants. Call recording with audio player. QoS, firewall and NAT traversal, VPN/ UDP/TCP and SRTP/TLS support. Configuration and remote provisioning (OEM version). % For more visit iscoord.com. For more visit broadvox.com. SIP Ad B Final.ai 1 1/24/ :23:46 AM SECURING THE SIP TRUNK Safe SIP Trunk Adoption without Toll Fraud or Security Risk Thinking about SIP Trunking? Worried about Security? What about Survivability? How about Interoperability? Comprehensive SIP Trunk solution: C Enablement: Terminate SIP Trunks for enterprise communications... seamlessly, safely M Y CM Control: Enforce granular policies... manage users and access in real time N ew MY CY CMY Protection: Guard against application-layer threats including toll fraud, eavesdropping and unauthorized access Enterprise Session Border Controllers (E-SBCs) from AudioCodes. K The industry's only Common Criteria certified device for SIP Trunk termination Download Dow wnload Sipera s white paper MediantTM 800 E-SBC MediantTM 1000 E-SBC Securing Se SSecurin ecurin ecur curing tthe he SSIP IP TTrunk runk ww ww.sip MediantTM 3000 E-SBC Copyright 2011 Sipera Systems, Inc. All rights reserved. Sipera, SLiC, Sipera UC-Sec and related products, Sipera LAVA and Sipera VIPER are trademarks of Sipera Systems, Inc. For more information:
6 Winter/Spring Telecom Reseller: SIP Report SIP Guide Company Solution/Service name Description Contact Contact 360networks Wholesale VoIP - IP origination, Our VoIP network is 100 percent IP-based and runs on our wholly owned 18,000-mile fiber-optic backbone, Steve Cardwell termination, toll free providing cost-effective, reliable, and deep coverage throughout the western United States. 3CX 3CX Hotel Module for 3CX The HM is an add-on for the 3CX Phone System, turning it into a full featured Hospitality PBX. The module adds Jaymes Marsh Phone System (HM) check in/out, Do not disturb, CDR/SMDR billing and more. 3CX 3CX Phone System for Windows 3CX Phone System for Windows is a software based PBX that replaces the traditional hardware PBX with an open Jaymes Marsh standard, manageable IP PBX. 3CX 3CX Voice Application Designer (VAD) VAD easily builds powerful voice applications that can improve customer service and employee productivity. Jaymes Marsh VAD s visual interface can put together a voice app, without having to spend weeks on code. 3CX 3CXPhone VoIP Phone for Windows 3CXPhone is a free softphone for Windows and has a smartphone interface which includes important business Jaymes Marsh features and allows the use of headsets. Save on electricity, hardware and administration. 3CX 3CXPhone for the iphone 3CXPhone for iphone is a Free SIP based VOIP Phone for iphone, ipad and ipod Touch devices. Works with 3CX Jaymes Marsh Phone System, Asterisk and popular VoIP/SIP providers. 3CX 3CXPhone VoIP for Android 3CXPhone for Android is a Free SIP based VoIP Phone for popular Android phones. Works with 3CX Phone System, Jaymes Marsh Asterisk, VoIP/SIP providers and leading Android phones. AudioCodes CPE Media Gateways Ideal for integrating existing TDM systems into SIP Trunking or new SIP IP-PBXs, CPE Media Gateways by AudioCodes convert SIP protocol communications to legacy formats including analog FXO or FXS; digital BRI, PRI or CAS T1/E1 formats. For more information visit: AudioCodes Enterprise Session Border Controller AudioCodes family of Enterprise Session Border Controllers enables connectivity, security and survivability for (E-SBC) enterprises looking to migrate from TDM to SIP Trunking services. For more information visit: AudioCodes High Availability Media Gateways AudioCodes line of carrier-grade high availability media gateways for service providers and enterprises are a reliable and high quality solution that can interconnect with market leading Softswitches, application servers, IP-PBXs and other standards-based VoIP elements. For more information visit: AudioCodes IP Phones AudioCodes 300HD Series of High Definition IP Phones offers a new dimension of voice call quality and clarity for the IP Telephony market, ideal for the Enterprise and Service Provider markets. For more information visit: Broadvox Go! SIP Trunking Products Broadvox offers a family of GO! SIP Trunking products that support unlimited calling and bundled service offerings, to satisfy the needs of businesses from SMBs and contact centers to Enterprises and carriers. Broadvox GO!Local The GO!Local SIP Trunk offers unlimited local calling within a calling area. GO!Local is priced at up to 70% less than traditional circuits. Discounted long distance or toll free minute bundles can be added. Broadvox GO!Domestic The GO!Domestic SIP trunk consists of discounted bundled long distance and toll-free minutes, as well as unlimited concurrent call sessions. Go!Domestic can be shared across multiple locations. Broadvox GO!Anywhere GO!Anywhere SIP trunk offers unlimited local and long distance calling with discounted international calling. GO!Anywhere interoperates with an extensive list of business Unified Communications solutions. Broadvox GO!Broadband GO!Broadband is the Broadvox offering of connectivity for SIP Trunking customers - via DSL, T1, and ethernet over copper connections. Coupling GO!Broadband with GO! SIP Trunking products, secures QoS throughout your network. Broadvox Unified Communications as a Solution: Broadvox offers hosted unified communications platforms to SMBs and Enterprises, whose needs vary from basic GO!VBX, C3 IP & C4 IP IP PBX functionality to advanced integrated UC capabilities. GO!VBX supplies SMBs access to hosted services at a substantial savings - only $10 per extension (Save up to 60% over other hosted offerings). C3 IP and C4 IP offer a full suite of hosted services and tailored solutions, for larger businesses and Enterprises. Cordia Corporation CordiaIP VoIP and SIP Trunking Cordia is a leading global communications provider of VoIP and SIP technologies. Cordia s world class Enterprise Dave Coughlin Solutions products include Cordia SIP Circuit, SIP Voice On Demand (VOD) and SIP Power. We mean business. Global Crossing SIP Trunking Solutions; Global Crossing s enterprise SIP trunking solution delivers a rich breadth of services with carrier-class quality, James Harney siptrunking Global Crossing VoIP Outbound, reliability and security for maximum savings on overall telephony costs, reducing total cost of Global Crossing VoIP On-Net Plus, Global Crossing VoIP Toll Free and Global Crossing VoIP Local Service Interactive Intelligence CaaS Contact Center CaaS Contact Center is a comprehensive set of SIP-based on-demand services for contact center automation. Sales/Marketing Inc. Interactive Intelligence Customer Interaction Center (CIC) CIC is a SIP-based all-in-one communications software suite that provides multichannel contact center automation Sales/Marketing Inc. and enterprise IP telephony functionality for mid-size to large organizations. Interactive Intelligence Interaction Process Automation (IPA) IPA is an all-in-one SIP-based process automation solution designed to prioritize, route, escalate and track each Sales/Marketing Inc. step of a business process. IPsmarx Technology SIP Trunking Solution By taking advantage of the IPsmarx SIP Trunking Solution, origination providers can deliver SIP-based DID numbers, Carrie Fedders also known as virtual phone numbers, to VoIP Service Providers, Call Centers, Corporations, and Calling Card Operators.
7 Telecom Reseller: SIP Report 7 Winter/Spring 2011 SIP Guide Company Solution/Service name Description Contact Contact iscoord ag is-phone Assistant for Broadworks is-phone Assistant for BroadWorks is an add-on for the is-phone plug-in for IBM and is-phone Portable. Chris Peter Add-On All BroadWorks features are implemented and enable full CTI functionality with one click. iscoord ag is-phone communicator UC is-phone communicator is a white-label OEM softphone integrated into Microsoft UC platform (Exchange/Outlook/ Chris Peter Client for Outlook Live Messenger), enabling audio/video calls or conferences with any SIP based PBX/IMS platform or SIP carrier. iscoord ag is-phone for iphone/ipad/ipod Touch The is-phone for iphone/ipad/ipod Touch is a softphone with enterprise-grade features incl. conference-calls, Chris Peter interoperable with IP PBXs/IMS platforms/sip carriers. Available in the itunes App Store and offered as OEM product. iscoord ag is-phone Plug-In for IBM Lotus Notes is-phone for IBM Lotus Notes/Sametime is a feature rich softphone plug-in integrated with the IBM UC platform, Chris Peter 8.0.1/8.5 and Sametime 8.5 enabling audio/video calls/conferences with any SIP based PBX/IMS platform or SIP carrier. iscoord ag is-phone Portable Stand-Alone UC Client is-phone Portable is lightweight stand-alone UC client, enabling audio/video calls/conferences as well chats with Chris Peter any SIP based PBX/IMS platform or SIP carrier and is easy to install and operate. iscoord ag is-phone Stand-Alone Client for IBM is-phone for IBM Lotus Notes 6/7/8 is a feature rich stand-alone softphone integrated with the IBM UC platform, Lotus Notes 6/7/8 enabling audio/video calls/conferences with any SIP based PBX/IMS platform or SIP carrier. Chris Peter Legacy SIP LegacySIP Ascend 24 (LSA24) Our SIP trunking gateway, the LegacySIP Ascend 24 (LSA24) sits in front of your PBX, between your PBX T1 port and Claude Hayn the Internet. As soon as you power up the LSA24 it automatically configures itself and immediately allows you to make and receive VoIP phone calls using our low-cost SIP trunk service. The LSA24, allows users to seamlessly use SIP trunks to make and receive VoIP phone calls over the Internet. Seamless T1 and PRI to SIP conversion, Seamless SIP to T1 and PRI conversion Level 3 Communications SIP Trunking SIP enabled local and long distance voice service delivered to Enterprise UC and PBX platforms to enable voice Jason Brougham communications in medium to large businesses. Net2Phone Net2Phone SIP Trunking Net2Phone enables partners to deliver high-quality, low-cost voice services to their customers who use Asterisk and Jonah Fink other SIP-based IP-PBXs and softswitches. Simple Signal SimpleSip SIP Trunking solutions SimpleSignal is a facilities-based Unified Communications provider of business VoIP and SIP Trunking. It is certified David Gilbert to bring Sip Trunking to Avaya,Cisco, Digium and many others. SimpleSignal services are deployed throughout the siptrunking.php United States and 6 foreign countries. Find us on the web at Sipera Sipera UC-Sec The Sipera UC-Sec offers comprehensive, plug-and-play unified communication (UC) security solutions enabling Andy Asava enterprises to address security concerns and define security boundaries while enjoying the benefits of SIP trunks and UC. SoTel Opti-Flex IP Measured service, Enterprise pricing for small business customers, Unlimited concurrent calls, Virtual telephone Jason Brougham number availability, Flexible number routing, Totally E-911 compliant, Maximum cost savings SoTel Opti-Max IP Unlimited local and long distance service at a flat rate, Includes white pages listing, operator services, and inbound Jason Brougham caller ID name and number, One concurrent call per SIP service line, Fully E-911 compliant, Uses existing or optional bandwidth service, Maximum cost savings SoTel Opti-Max IP Features: Unlimited local and long distance service at a flat rate, Includes white pages listing, operator services, and inbound Jason Brougham caller ID name and number, One concurrent call per SIP service line, Fully E-911 compliant, Uses existing or optional bandwidth service, Maximum cost savings TalkSwitch Corp Phone Systems for Small Business TalkSwitch IP PBX phone systems are designed for small business - they re easy to use, affordable and reliable. Sales: TalkSwitch systems include everything small business needs to sound like big business x301 There s More! On these pages are featured products and services in a growing SIP universe. This our first of a series of guides on SIP. Our next guide this spring will be on end points, devices and other SIP ready gear. You can also find news continuously updated at To be listed or to learn more please contact Marie-France at or call CALL ACCOUNTING
8 Spring Telecom Reseller Legacy SIP SIP for the SIP unready While SIP has become the dominant protocol for communication devices, it s easy to forget that the overwhelming majority of the installed base consists of products that were never designed with SIP in mind. Even as SIPready becomes the order of the day, the reality that almost every IT manager faces is that they will have one or more sites that are completely SIP Unready. LegacySIP.com was created with this in mind. LegacySip.com enables enterprises and SMBs that are operating legacy equipment to take advantage of the savings and other improvements that SIP provides without having to forklift upgrade their current platform or end points. They don t have to do anything, said Claude Hayn of LegacySIP.com. No network or infrastructure updates are needed for us to get you SIP now! Small businesses with one or two T1 PBXs are in a perfect position to benefit from SIP savings, but are often unable to do so because of the costs involved in migration. In addition to dramatic cost savings SIP is the backbone of future communications. Embracing SIP now will allow companies to take advantage of these business solutions. Although SIP is currently best known for voice communications (VoIP), the future is for SIP devices that facilitate numerous business communications, including enterprise level presence, instant messaging, chat, and video using SIP as the common protocol. We have created a best-of-breed infrastructure that effectively allows companies with legacy PBXs to easily begin their migration path to a SIP based infrastructure, says Hayn. Our customers can have any type of legacy infrastructure. We do not need to add any additional cards or software to their PBX in order for them to benefit from using our SIP trunks. Our highly customized gateway resides at the customer site, connects to the Internet, interacts with our soft switches directly and translates all call traffic to T1/PRI format. Our SIP trunking gateway is Plug-N- Play, and normally takes about 30 minutes to install SIP trunking availability. Usage begins immediately. Users simply take their T-1 line and plug it into our box. Everything else is transparent and automatic. We manage everything else through our highly customized, secure gateways. This allows us to respond to customer needs very quickly, with confidence and well-trained individuals. Instead of forcing companies to take T1 capacity our customers choose their individual usage model based upon any combination of actual minutes and concurrent calls. LegacySIP.com offers additional capabilities like presence awareness and instant messaging that our customers can take advantage of. Remote workers are able to connect to and use our customer gateways and all related services. Hayn says, We anticipate that our legacy PBX customers will add SIP devices as they need additional or replacement handsets. At some point over time customers Ad will Layout1.fh11 reach a critical 4/26/06 mass 2:51 PM of SIP Page devices 1 and likely migrate to an ip PBX. Conveniently our SIP Gateway is also a full-blown ip PBX. There will be no need for customers to acquire an additional ip PBX solution, they will already have one. LegacySIP.com is currently in development of a standalone conferencing solution that customers can use without consuming their Legacy PBX capacity. This standalone conferencing solution integrates with our SIP Gateway and runs independently of the customer PBX. They are also building mobile phone solutions that function as PBX extensions. Our roadmap is to build upon our C M Y CM MY CY CMY K relationships offering our customers the ability to pick and choose additional SIP related technologies to increase their business response capabilities. % For more visit Composite