Technical Jargon

VoIP Technology In A Hosted Environment
Next Generation Telecom Network

Kayote Networks develops the world's most advanced hosted VoIP platform designed for tomorrow's Next Generation Telecom Networks. Built to offer the highest quality carrier-to-carrier communications connectivity worldwide while automatically resolving the most complex VoIP challenges of interconnectivity, interoperability, and security.

VoIP Session Control Management

Kayote's Next Generation Telecom Network concentrates on the session control and management aspects within the overall architecture of VoIP. Its enhanced services and functionality resolve a multitude of interconnectivity and interoperability issues while ensuring a secure call flow.

The following diagram provides an overview of the Kayote's Next Generation Telecom Network features.

Hosted VoIP Feature Diagram

Integrating a Hosted VoIP Solution

Kayote's hosted VoIP platform has been integrated with numerous partners and is available to companies of all sizes as a complete solution, the advanced VoIP Traffic Management System (VTM). As a "hosted switch" that handles all the call traffic of a carrier in a secure and hosted environment, VTM transmits any combination of voice, video, and data, letting companies manage their additional carrier-essential information.

VoIP Interconnectivity

Kayote provides its customers with the ability to painlessly interconnect with almost any terminator or device. This invaluable service facilitates fast interconnects thus enabling carriers to build their businesses and maximize revenue. Kayote understands the importance and complexities of managing interconnects, which is why we have simplified the process so that virtually anyone can now setup new routes.

Our hosted VoIP Traffic Manager (VTM) simplifies the multifaceted issues of interconnecting, as well as ensuring that once routes are in place they can be properly monitored and maintained. If there is a problem, the system can send an alert and automatically direct call traffic to any other high quality route you designate. The following are a few key features our system offers to help you guarantee your call quality:

• Adaptive and quality-based LCR, including traffic shaping and route throttling
• Online analytical processing, providing route statistics of aggregate and averaged traffic parameters, cost analysis, and quality
• Robust and fully redundant, globally distributed VoIP network providing carrier grade call quality and high availability
• Multi-layered VoIP security, including VoIP firewall capabilities and intruder prevention, SPIT (Spam for Internet Telephony) control, and secure ENUM functionality

VoIP Interoperability: Protocol Conversion and Cleanup

Sometimes, true interoperability between VoIP entities can only exist with third-party intervention. Companies quickly realize that that even under standards-based conventions for VoIP protocols, such as SIP and H.323, carriers and ITSPs cannot automatically terminate between each other due to a range of protocol conflicts. Even different versions of the same protocol often prevent successful termination of calls and may even crash telephony networks. This inconsistency has resulted in millions of dollars of lost opportunities and tremendous customer apprehension.

Enabling Seamless Peering

Kayote's technology ensures seamless conversion between SIP and H323, enabling even the largest VoIP carriers, who would otherwise be incompatible to terminate traffic with each other and bridge their technology gap. Our interconnectivity solutions generate new and highly profitable partner relationships.

Real-time Understanding of Potential Protocol Conflicts

Incompatibility can prove potentially fatal to a telephony carrier's business. Kayote's internally developed systems incorporate back-to-back user agents providing real-time handling of potential protocol conflicts. Our hands-on experience with actual peering scenarios provides the ultimate flexibility to solve new peering problems. To maintain the highest quality and uptime of our systems, we are constantly testing both hardware and software used by telephony carriers, analyzing telephony traffic records, and resolving other strange VoIP conflicts.

VoIP Interoperability: NAT Detection and Traversal

Only by determining the correct NAT (Network Address Translation) environment of each VoIP end-point can the voice stream pass correctly between them. With an increasingly large number of enterprises implementing VoIP systems inside their corporate LANs, NAT detection and traversal will be critical to making or breaking a successful VoIP deployment.

Intelligent NAT Traversal

Kayote's technology allows you to determine the most efficient method to pass the RTP (Real-time Transport Protocol) voice stream between end-points. Our proprietary RTP media relay will seamlessly takes over the media for any NAT environment. In most situations, our built-in NAT detection and traversal algorithm automatically ensures the highest quality level of the RTP stream.

The Most Cost Effective Alternative to Expensive Equipment

Kayote offers the ability to proxy media thru our proprietary RTP media relay. By placing the proxy at the customer's premises, we provide the most cost effective alternative to expensive equipment.

VoIP Interoperability: Codec Conversion and Enforcement

In the VoIP environment, different codecs are used to encode and decode the analog signals of the voice stream into the binary and digital environment of the Internet. By using different codecs based on the type of software and hardware, protocol, and available bandwidth for each call, VoIP operators often cause call incompatibilities.

Bridge the Codec Restriction Gap

Kayote's technology manages the codecs on both ends of the call, regardless of the differences between them. This ensures call reliability and increases the available universe of termination partners.

Kayote understands that both business and technical considerations often require a company to select a particular codec. We automatically enforce these preferences, overriding alternative codecs that could potentially cripple communications.