Meru Wireless Backbone System
The Meru Wireless Backbone System differs from most other wireless networking systems by using a ï¿½blanket,ï¿½ in which every access point uses the same 802.11a and b/g WiFi channels. It essentially aggregates all the GHz frequencies together to give you more bandwidth ï¿½ an aggregate of 240 MHz of backbone spectrum. This product aims to replace the wire going to the access point (AP) and uses wireless instead, essentially making this ï¿½zero-wireï¿½ to the desktop and the backbone. This solution enables an enterprise to go all-wireless. The WLAN is the only network ï¿½ from the core to the device. Meru Networks claims that this product is the only one in the market that can support both voice and data over multi-hop wireless backhaul, with a switch-like experience over an all-wireless enterprise backbone, essentially making them the only solution to ï¿½unwireï¿½ end-to-end for both voice and data.
Meru Networks told TMC Labs, ï¿½In the future, all networks will work this way. Wireless CATx will be the de facto connectivity mechanism. The industry segments that are first adopters of this product are: verticals such as universities, manufacturing, retail, hotspots (airports, convention centers, etc.) Horizontal targets are in geographies where old buildings are not pre-wired. This entails large markets in Asia including China, India, Japan, and the remaining Pacific Rim.ï¿½
The market for this type of innovative product will be especially strong in the SMB, which will enjoy very low deployment cost and no radio frequency (RF) expertise requirements, with the added benefit of the elimination of wire for both voice and data, as well as easier moves/adds/changes.
The Meru Wireless Backbone System provides hierarchical aggregation of capacity over the air similar to EtherChannel, and the ability to provide multi-hop QoS and end-to-end security, and the ability to provide link redundancy and dynamic self-healing.
Meruï¿½s WLAN product claims to be the first to enable the ï¿½unwired officeï¿½ ï¿½ entirely cutting the access cable for voice and data. Osaka Gas, one of Japanï¿½s largest utility companies with 50 offices and 6,000 employees, went entirely wireless ï¿½ no wire for voice or data in any office barring disaster recovery ï¿½ last year.
Some other unique features include full-duplex wireless channels, a significant innovation based on patented antenna technology; Multi-hop QoS for commercial-grade voice+data support, extending Meruï¿½s Air Traffic Control access technology to the wireless backbone; Seamless handoff and Virtual Cell, extended over multi-hop wireless backbone. If the clients are on 802.11g, the system can use the 802.11a channels for the backbone. Instead you can place all the data clients on one 802.11g channel and use the other channels for voice, and the ï¿½aï¿½ channels for the backbone.
When 802.11n arrives, giving more capacity, the system will be compatible with it. Supporting both voice and data over a multi-hop wireless network is unique in the industry and, Meru believes, the pioneer of a trend that will eventually replace wire with wireless entirely. While weï¿½re not ready just yet to give up our super fast Gigabit-wired Ethernet connection, in favor of a 100 percent wireless solution that is slower, we admire Meru for their vision of an unwired world.
NICE Systems Ltd.
NICE Contact Center Interactions Solution, VoIP Enhancement
NICE has been one of the leading call recording and monitoring companies in the world has even diversified their call center feature/function portfolio by adding workforce management with their Performix acquisition and recent IEX acquisition. NICEï¿½s next-generation VoIP active recording solutions, with specialized contact center functionality, enable efficient management and administration of branches and agents from home and enhance centralized recording capabilities for distributed environments. By eliminating the need for setup, administration, and management of recording hardware at branches or remote sites, IP-based active recording profoundly reduces the overhead and complexity associated with previous generation VoIP recording. NICEï¿½s VoIP active recording solutions enable efficient and centralized management and administration of branches and agents from home.
The new offering is the first to include solutions for active recording in Nortelï¿½s newly developed Duplicate Media Stream over IP (DMS-IP) architecture, active recording for Ciscoï¿½s CallManager, as well as offer new redundancy options for Avayaï¿½s Communication Manager API, and IP-phone applications. Nice also supports passive monitoring and includes support for all the major IP telephony vendors, including Alcatel, Aspect, Avaya, Ericsson, IPC, Mitel, NEC, Rockwell, Siemens, and others.
The nice thing about NICEï¿½s VoIP recording solutions (pun intended) is that they are an integral part of the companyï¿½s unified product architecture and suite of solutions. This means a smooth migration to VoIP that is transparent to the user, thus providing investment protection. NICE also offers hybrid solutions that cover the entire spectrum of customer interactions management, combining traditional and VoIP, liability, and quality monitoring for contact centers. NICE offers software-only, scalable VoIP solutions that are certified by the worldï¿½s leading VoIP switch vendors, addressing small-scale to large, multi-site, high-end environments. Finally, NICE offers speech analytics that can perform word spotting and even determine customerï¿½s state of mind (i.e., angry, excited) to aid in agent training and improve the customer relationship.
BLC 6314 10GigE Transport and Optical Line Termination (OLT) Blade
Occamï¿½s razor has always been associated with the concept of simplicity. For instance, Occamï¿½s razor postulates that the answers to most scientific phenomenon are usually the simplest rather than the more complex. Well, Occam Networks aims to provide the simplest solution to the complexity of bandwidth hungry applications running on a service providerï¿½s network. Occamï¿½s newest blade for its flagship BLC 6000, the BLC 6314 10GigE Transport and Optical Line Termination (OLT) blade, adds a whopping dual 10 Gigabit Optical Ethernet and multiple 1 Gigabit Ethernet interfaces to the BLC 6000 in a single, environmentally hardened blade. The Occam BLC 6314 claims to be the first and only product that delivers this level of bandwidth with a sub-50-millisecond failover rate. This enables telcos to create highly resilient access networks that can deliver a TV signal with ï¿½five ninesï¿½ of reliability.
The BLC provides an abundance of symmetrical bandwidth that enables telcos to easily create very high capacity access networks with 10 Gigabit Ethernet redundant transport rings at a fraction of the cost of traditional SONET-based OC 192. The BLC 6314 can be used for fiber to the home (FTTH) optical line termination. When combined with the Occam BLC 6312 blade, the BLC 6314 creates a high-capacity FTTP Optical Line Termination system with integrated 10 Gigabit Ethernet transport that can serve more than 250 subscribers with up to one Gigabit of full-duplex bandwidth each. Having this kind of mega-bandwidth makes broadband cable and DSL look like dial-up!
Also, the BLC 6314 can also be used for higher capacity access network aggregation using the companyï¿½s Ethernet Protection Switching (EPS). It provides resilient 10 Gigabit Ethernet aggregation rings for BLC 6000-based copper and fiber access networks. As bandwidth-intensive applications, such as HDTV, IPTV, VoIP, VoD, P2P, etc., continue to consume more bandwidth, service providers are looking for solutions to their subscribersï¿½ seemingly insatiable bandwidth appetite. Fortunately, Occam Networksï¿½ BLC 6000 Broadband Loop Carrier (BLC) uses IP as a simple, common service delivery protocol for all services and Ethernet transport to provide the economical, highly scalable bandwidth needed for delivering todayï¿½s and tomorrowï¿½s new services.
Paragon Wireless Inc.
The ability to seamlessly make and receive phone calls on both cellular and WiFi networks is not a dream for tomorrow ï¿½ itï¿½s happening today with companies like Paragon Wireless leading the way. The Paragon Wireless PWTW-1100 is the first commercialized SIP-based, voice optimized, dual mode (GSM/WiFi-VoIP) handset in the world. In fact, they started shipments in China in March 2006.
Paragon Wireless has developed a patent pending technology that allows an optimal integration of two wireless radios (cellular and WiFi) in such a way that problems like power management, roaming, FMC, Mobile VOIP and security (WPA supported) are solved. It can automatically roam between radio networks in less than 50ms.
Paragon has created the first commercialized product in its segment and the technology solution of integrating WiFi and VoIP with cellular radio in a way that also allows great performance. Paragon claims that independent tests have been conducted by China Telecom testing standby and talk times of their dual mode phone to confirm its good performance. The handset sports four hour talk and 72 hours standby time, even when both its WiFi and GSM capabilities are running. It has strong VoIP support with acoustic echo cancellation, and a suite of codecs that includes G.711, G.723, AMR, and LPC, as well as a dynamic jitter buffer and a voice activity detector.
The phone features the most common PDA applications, including calendar, schedule manager, alarm clock, voice recorder, and more. It also features IM, e-mail, an Internet browser, and a VPN client. With its built-in SD memory card slot and MP3 player you can play music as well. It utilizes Intelï¿½s PXA271 processor, a 2.4-inch TFT touch screen (320x240 QVGA, 260k colors, and backlit), a built-in speaker and microphone, and a 1.3 megapixel CMOS camera. The phone includes a simultaneously active 802.11b interface and a tri-band GSM/GPRS interface. The product is a normal cell phone size handset with large TFT display. Itï¿½s worth mentioning that both radios use the same user interface. While in WiFi call the user can receive cellular call and switch between the calls if needed.
As enhanced applications converge onto IP networks and as service providers look to deploy Triple and Quadruple Play services, the ability to diagnose and troubleshoot network issues becomes even more critical and more complex. Fortunately, RADCOMï¿½s R70 provides what you could consider a ï¿½triple play packet sniffer.ï¿½ The Linux-based, intrusive and non-intrusive R70 probes monitor and analyze network traffic. The R70 probes are a high-end, upgradeable component of RADCOMï¿½s Omni-Q, a network and service monitoring, analysis and troubleshooting system for public IP, cable, and advanced cellular networks. R70 probes give operators total visibility into the session and application levels, with full seven-layer analysis. They provide operators with a comprehensive and correlated view of all IP-based services, including VoIP and IPTV, complete with service integrity metrics from their subscriberï¿½s perspective.
The R70 is the first probe in the market to process data at 10 Gbps. The R70 leverages the companyï¿½s existing, proprietary technology (the Gear Chip) to achieve its high level of performance that exceeds the processing limitations of most of not all other existing probes in the market. It is a unique monitoring and analysis tool that, in a single product, addresses the needs of ILECs, MSOs, and cellular operators.
Service providers moving to an all-IP core network architecture will find that RADCOMï¿½s R70 offers a comprehensive feature set to monitor and analyze their networks. Because it gives operators complete visibility in the IP-based service running over the network, it enables early stage fault detection, pre-emptive maintenance and optimization. Additionally, when operators seek to introduce new services to create new revenue streams, RADCOMï¿½s R70 probe-based monitoring system reveals subscriber behavior providing valuable insight into the new service adoption process.
RingCentral Online 3.0
RingCentral Online 3.0
The ability for customers to reach you is important for customer retention. Large corporations have implemented many technologies to that end such as ACDs, digital fax servers, and more. Unfortunately, small businesses or entrepreneurs often cannot afford these
systems. RingCentral aims to address to provide advanced contact technology to small corporations with 10 or fewer employees, mobile workers employed by larger firms, and road warriors. Small business customers include owners of home-based businesses, consultants, non-profit organizations, and the like.
RingCentral provides advanced telecommunications services, unifying landline, wireless, VoIP and e-mail communications, and even exceeds services available to large enterprises with a fixed PBX. RingCentral provides dial tone independence, full fax integration, and does not require customers to purchase any new hardware or software ï¿½ the service works seamlessly with existing landline and mobile phones. RingCentral gives you the ability to manage messages, faxes, and call records over the Web, on a PC, and over the phone
RingCentral Online combines a toll-free and/or local number with advanced call management, PBX, voicemail, and Internet fax, enabling customers to automatically screen, forward, and place calls, take voicemail, send and receive faxes, and receive message alerts.
RingCentral utilizes their Call Controller tool, a desktop pop-up window that allows customers to react to calls in real time without having to touch their phone. The Call Controller toll gives customers four clickable choices for handling the call: accept, reject, send to voicemail, or reply with a text-to-speech response. This text-to-speech technology is a very innovative feature. Essentially, while customers are on hold, you can type a message, which will be translated to speech and broadcast to the caller.
RingCentral has powerful answering rules including call scheduling, filtering, and routing technology, which allows customers to greet callers with various messages and route them according to the day, time, date range, and caller ID. Customers can route calls to voice mailboxes, extensions, and any phone in the world, as well as block calls with exceptions, allowing users to set specifications and overrides for VIP customers or family members.
Their latest feature, RingMe, letï¿½s customers embed a button into their Web sites or e-mail signatures so that online visitors can reach them by phone with click-to-call technology. Additional new and enhanced features include FaxOut for sending faxes from any computer, Fax From E-mail, and Caller Preview function, whereby customers can hear whoï¿½s calling before deciding what to do with the call. RingCentral also features integration with Microsoft Outlook and Outlook Express.
A200 FXO/FXS Analog Telephone Support System
TMC Labs has tested Sangomaï¿½s cards firsthand, so we know how versatile and innovative these cards are. Sangoma is well known for architecting their telephony hardware to support several softpbxs, including the popular Asterisk platform, Yate, OPAL PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications.
Sangomaï¿½s A200 and Sangomaï¿½s REMORA system together comprise the FXO/FXS version of the companyï¿½s Advanced Flexible Telecommunications (AFT) hardware designed for optimum support of analog voice traffic. The A200 series provides analog connections for FXO or FXS in expandable solutions from two ports to 24 ports.
One of the most unique aspects of all of Sangomaï¿½s products is that they are easily expandable to go from two ports to 24 ports utilizing the existing card and adding modules. This type of flexibility and field upgradeability has never been seen in this space before. Another innovative feature that Sangoma introduced in many of their products, including the A200, is a telco-grade hardware echo canceller of 128ms.
The A200 solution supports any combination of up to 24 FXO or FXS connections. A single PCI slot host connection for all ports ensures common synchronous clocking for all channels. The base AFT architecture is shared with Sangomaï¿½s A101, A201, A104, and soon to be released A108 cards, ensuring common 3.3V/5V, high performance PCI compatibility. Just as with all Sangomaï¿½s cards, the A200 has field upgradeable firmware to take advantage of hardware and software improvements as they become available.
The A200 consists of a REMORA daughterboard mounted on the AFT PCI card. The REMORA card has two sockets, each of which can accept a FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS ports respectively. Up to five additional REMORA daughterboards can be mounted in empty slot positions beside the A200 assembly connected to the A200 by a backplane bus connector. Importantly, the A200 fits into the 2U Form factor and short 2U compatible mounting clips are available for installation in 2U rackmount servers. It also features a 32-bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Finally, it is fully PCI 2.2 compliant, making it compatible with all commercially available motherboards, and it performs proper interrupt sharing ï¿½ something that often plagues other telephony hardware.
The ShoreTel system is a fully distributed IP phone system with no single point of failure, making it a reliable solution for multi-site enterprises. Created specifically to run enterprise-class telephony and voice services across multiple sites, ShoreTelï¿½s systems do not suffer from the complexities of traditional systems evolved out of legacy voice or data switches.
The ShoreTel system offers distinct benefits because of its unique, distributed architecture. For instance, call control is distributed to voice switches and voice applications, including voicemail and automated attendant that run on standard server hardware anywhere on an IP network. There is no single point of failure and there is a single system image across all geographies with complete feature transparency, making the system very easy to manage.
ShoreTelï¿½s switches and software are designed to provide easy deployment, and work well with their ergonomic and feature-rich ShoreTel IP phones. In fact, their latest phone is the ShorePhone IP 212k, an ergonomic IP key system telephone with 12 programmable buttons, exceptional audio quality, and big LCD display. Other features debuting in the latest ShoreTel release include a gigabit IP phone, the IP 560g, a new staff IP phone, the IP 230, Centrex flash capabilities, personal-assistant and Caller-ID enhancements. The ShoreTel system includes comprehensive key system functionality and it allows bridged call appearances to span locations, thus, virtualizing key system behavior across the enterprise.
One of the most unique aspects of ShoreTelï¿½s product line is that itï¿½s the same exact hardware, whether you are 10 employees or 10,000 employees ï¿½ a simple software license increases the number. A single hardware platform means less training for the reseller and VAR channel as well as easy upgrades. Competitive solutions include distinct products for different sizes of companies, forcing folklift upgrades when customers move between platforms and impacting customer service levels by requiring partners and customers to support multiple, complex products.
Sipera IPCS 310
Edwin Andres Pena made headlines when he hacked into VoIP providersï¿½ networks, stole $1 million worth of VoIP minutes, and then resold them. This is even more evidence that intrusion and detection systems (IDS) specialized to handle VoIP are needed in our industry. The Sipera IPCS 310 is a comprehensive IP Communications Security appliance that includes a complete suite of security features for protecting session-based, real-time IP communications applications including VoIP, IM, multimedia, and other collaboration tools.
It can be deployed within the enterprise network in front of communications servers, between the voice and data VLAN, or along the SIP trunk, to prevent any potential attacks in real time. The Sipera IPCS 310 protects both the infrastructure and end users from a number of malicious, application-specific attacks and service abuse from DoS/DDoS flood and fuzzing attacks to more sophisticated reconnaissance, stealth, and VoIP spam attacks.
Sipera addresses all the required security functionality, such as firewall, IDS/IPS, DoS prevention, network level correlation, and spam filtering, while implementing sophisticated techniques to ensure unique VoIP threats are proactively recognized, detected, and eliminated. These techniques include anomaly detection and behavior learning. The Sipera IPCS 310 continuously learns call patterns and endpoint fingerprints, in addition to being able to constantly analyze raw event data and take automatic action, which gives the security solution the ability to evolve and adapt to effectively counter any threat. This level of sophistication is the only way to minimize false negatives (allowing bad calls) and positives (blocking good calls). This level of sophistication is the only way to identify both stealth attacks and VoIP spam, which are difficult to detect, as the real-time nature of VoIP does not allow the security system the luxury of storing the call while itï¿½s analyzed, before sending it on, as is
the case with e-mail or other non-time sensitive IP applications.
Additionally, the Sipera VIPER lab, comprised of top wireless and VoIP vulnerability research experts who proactively identify SIP, UMA, and IMS vulnerabilities, has catalogued thousands of threats that can be launched against these networks to date. This expertise forms the foundation of the Sipera IPCS 310 that protects all VoIP elements from a variety of malicious attacks and service abuse.
With dual mode phones and high-speed Internet connectivity (e.g., EV-DO, WiFi) on mobile phones, customers are looking for to leverage softphones on their mobile phones for inexpensive VoIP minutes and more ï¿½connectedness.ï¿½ TeamSpirit Mobile empowers OEMs with powerful VoIP functionality, bringing softphone applications to mass-market mobile devices. This product makes it easy for the equipment manufacturers and mobile software developers to build feature-rich, easy-to-use softphone applications. These innovations will allow the end users to quickly adopt wireless IP capabilities of their mobile phones and enjoy the benefits of VoIP communication. TeamSpirit Mobile also includes a Voice Engine Software Development Kit (SDK) for mobile devices to help equipment manufactures develop high performance softphone applications.
Spirit DSP claims that TeamSpirit Mobile is the fastest voice engine running on mobile devices today. They also claim that their mobile VoIP engine is the industryï¿½s first to run a VoIP application on devices with CPU clock rate of 200 MHz or less, for example, the Qtek 8310 and the iPAQ 6340.
TeamSpirit Mobile addresses the most challenging problems encountered in mobile IP applications, such as ensuring rich voice and quality video while dealing with limited system resources of a mobile device. This solution is optimized for appliances running under Windows Mobile 5.0, Windows Mobile 5.0 SmartPhone Edition, Windows PocketPC 2003, or Windows Smartphone 2003 operating systems. TeamSpirit voice engine has been specifically optimized for a dozen mobile platforms: OMAP 850, 1510, 1030 and others; Intel Xscale family; and any ARM9 processors starting from 168 MHz.
SPIRITï¿½s proprietary technologies deliver exceptional voice quality and eliminate echo and surrounding noises while utilizing minimal resources. Their voice engine also includes optimized codecs, PLC, voice enhancement components (Full-duplex Acoustic Echo Cancellation, Automatic Gain Control, Noise Suppression) and adaptive jitter buffer. It also supports a built-in or third-party SIP stack, provides network support (RTP/RTCP) and call control API, and integrates network and voice processing subsystems. It can also be enhanced with video support using standard H.263 and H.264 video codecs. Finally, since it has low resource consumption, it prolongs battery life on mobile devices.
InContact is a Customer Interaction Management service that combines multimedia customer contacts (voice, chat, e-mail, fax), skills-based ACD for both in-house and at-home agents, and a robust and completely customizable IVR. InContact is a redundant, hosted solution that leverages multiple core network carriers for maximum uptime. The product line addresses the needs of companies seeking to improve their customer contact experience by offering an integrated solution that addresses all customer interactions, including voice, e-mail, chat, and fax. Since the application is hosted, it grows as your organization grows and can be done very quickly. In fact, UCN was involved with quickly setting up the largest Spanish-speaking call centers spanning four countries to assist in Global Telesourcingï¿½s Red Cross/Katrina fund raising efforts.
UCN is unique in that they bundle two critical service packages together, namely the contact handling service application and phone service. By providing a single vendor solution, UCN customers get a single bill and single point of contact to resolve issues. Essentially, they are both the carrier and the enhanced software service provider. UCN incorporates a robust visual drag-and-drop programming capability, allowing the company to customize the IVR and skills-based ACD to meet their specific needs on their own, without the need for professional services.
InContact is highly customizable and can easily make moves, adds, and changes on the fly using a Web-based management tool. You can also customize the management dashboard (displays of reports and statistics in real time). InContact also offers database integration with the leading CRM database programs.
UCN is one of the true VoIP innovators building a nationwide IP network that was one of the first national, commercial VoIP networks in the country ï¿½ going operational in 1997. The product was the first to deliver voice calls to the customer site via traditional TDM services, enabling the customer to keep their existing phone equipment yet take advantage of IP services ï¿½ but hosted within the network.
IP Web Center
Verizon, a traditional carrier, expanding into providing advanced call center applications, is more proof that carriers are moving beyond simply providing ï¿½dumb voiceï¿½ circuits. Verizonï¿½s IP Web Center is a network-based IP Contact Center offering that supports inbound/outbound telephony, Web call back, scheduled call back, Web chat, collaboration, e-mail, and fax.
Verizon Business married its Voice over IP service to its Contact Center service expertise to deliver one of the industryï¿½s first end-to-end contact center solutions. By combining the power of Verizon Web Center with the Verizon Voice over IP suite of services, businesses can reap the benefits of VoIP across their entire enterprise operations and achieve new capabilities and cost savings afforded by IP telephony. Some powerful features include unified messaging, ACD/IVR, quality monitoring, intelligence contact routing (across multiple media types), ability to create guest supervisors, pre-announced caller information (whisper announcements), enhanced statistical information, and remote agent capabilities.
Verizon Web Center and Verizon Voice over IP now share the same network infrastructure and customer premises equipment so Verizon Business can activate a wide range of IP telephony services, including IP Web Center, Hosted IP Centrex, IP Integrated Access, IP Flexible T-1 and IP Trunking, at a given company location.
IP Web Center is well suited for mid-sized businesses that want to expand the feature set of their existing contact centers. Since IP Web Center is a hosted solution that simply requires a phone and a broadband connection, large companies can also use the service for agents to make and receive calls anywhere in the United States, thus reducing the need to expand hardwired contact center facilities. Itï¿½s worth mentioning that companies are increasingly moving to the IP contact center, taking advantage of an IP network environment to become more flexible and realize infrastructure cost savings.
Verizon Businessï¿½s newly expanded pricing options let customers pay as they go for IP-enabled Web Center services. Businesses only pay a monthly per agent price, plus call transport fees and associated IP phone equipment costs. This works well for companies that need to increase agent levels during busy holiday times and then reduce them when normal call levels resume.
XConnect Global Networks, Ltd.
XConnect has a unique peering, settlement-free (zero-cost call), multilateral alliance. These benefits include more advanced call features, improved call quality and lower or zero call costs. These benefits are achieved by allowing members to deliver calls between their subscribers without involving unnecessary intermediaries and, where both call parties are VoIP subscribers, by delivering calls via pure end-to-end IP connectivity and completely bypassing the PSTN and its associated limitations and costs.
XConnect has created a globally distributed ï¿½plug and peerï¿½ network that they tell us is the largest ENUM directory and the largest peering network in the world. XConnect provides what they claim is the first and only carrier-neutral network that resolves the challenges of VoIP interconnection, including numbering, settlement, interoperability, security, identity, and privacy across a globally distributed network of VoIP peering points.
The fast growing XConnect Alliance offers multilateral settlement-free exchange of traffic between VoIP service provider members. Alliance members peer on a settlement-free basis and pass on Zero-Cost or reduced cost calls to their customers. Calls are delivered via end-to-end IP connectivity, eliminating the need to traverse the PSTN, which would otherwise incur per minute costs, reduce call quality (because of limited bandwidth and repeated data-voice-data signal conversion), and preclude sophisticated features such as three-way calling, video calling etc.
XConnect offers a peering solution that combines ENUM Directory Management with multi-protocol signaling interoperability (i.e., H.323, SIP, IMS, IAX, PacketCable, etc.) as well as unique VoIP security features, such as avoiding misappropriation of misuse of the numbering database and sophisticated methodologies to identify, detect, and prevent Spam over Internet Telephony (SPIT) and CallerID spoofing.
XConnect has implemented genuine Caller ID authentication, validation, and normalization in a VoIP environment (based on a number of SIP messaging fields including FROM, CONTACT, remote_party_ID, and p_asserted_ identity).
VoIP providers strongly prefer to have the ENUM numbering database available locally to facilitate speedy queries necessary to determine when calls can be routed to a peering partner. At the same time, they are concerned about their sensitive client numbering data being distributed around the world to competitors. XConnect has developed a Local Directory Server (LDS) deployed locally, while also preventing misappropriation, misuse, and data-mining of the ENUM database data. The LDS can perform thousands of queries per second while continuing to guarantee the confidentiality of each operatorï¿½s own ENUM data. IT