Features

March 2008

IP Telephony

Build up to best-of-breed IP telephony

Interoperable components can be combined to meet individual needs.

by Haim Melamed

When IP telephony was first introduced to the enterprise a decade ago, it incorporated the concept of decoupling the private branch exchange (PBX) software, phones, line cards, trunk connections and telephony applications. IP was the protocol selected to connect the system components, providing location independence, easy adds moves and changes, and lower cost of ownership.

 Although IP telephony was based on Ethernet and IP as the Layer 2 and Layer 3 protocols, and on RTP as the voice media protocol, the most significant protocol–the signaling protocol between the system components–remained non-standard. This created a barrier that prevented many enterprises from benefiting from the advantages of a moving to voice over Internet protocol (VoIP)–increased productivity and reduced operational costs.

Over time, session initiation protocol (SIP) emerged as the de facto signaling standard for VoIP. In the enterprise, nearly all IP telephony vendors came to support SIP as an alternative to a proprietary protocol, allowing users to create a true best-of-breed IP telephony implementation in the enterprise.

Developers of computer telephony integration (CTI) value-added applications, such as conferencing, messaging, IVR, contact centers and recording capabilities, are currently implementing standards-based communications with media gateways and media servers, instead of using API integration with telephony boards. This new ecosystem enables developers to break free from the old world of CTI and create open, platform-independent, interoperable software solutions for the enterprise.

An enterprise selecting a new telephony system for the organization can now take advantage of the same flexibility as previously experienced with IT systems, as all of the components are standards-based and interoperable using SIP. The enterprise can choose IP-based hardware and software solutions that scale to individual network needs, as well as the price-performance of the following specific components.

  • IP-PBX software includes software-only proprietary solutions, as well as software-only open-source solutions or a combined hardware/software proprietary solution. All are now based on SIP as the session control protocol.
  • IP-PBX hardware platforms can be off-the shelf servers running Linux or Windows, or alternatively, hosted inside other components of the system like the media gateway.
  • IP-phones can be purchased from any vendor manufacturing SIP phones and are available in many forms, types and price levels.
  • Media gateways can be acquired from any vendor, as long as the vendor maintains full interoperability testing with the chosen IP PBX vendor and has the features required to maintain VoIP quality in the network.
  • CTI applications can be integrated into systems using SIP and can originate from many different vendors providing best-in-class applications, including conferencing, messaging, IVR, contact center and recording.
  • Media servers for small-scale installations can be implemented in software inside the IP PBX platform. Media servers for medium-scale installations (hundreds of users) can be integrated into media gateway platforms, but for large-scale installations (thousands of users), this component should be purchased as a separate hardware-based platform.
  • IP and LAN switching infrastructure can support any IP telephony system, provided that it is designed to provide quality of service for VoIP applications.

Enterprises can now build an IP telephony system by selecting components from a wide selection of vendors, while still keeping their IP telephony networks open to new components and applications in the future.

Haim Melamed is the director of corporate channel marketing at AudioCodes, Chicago.

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