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Understanding Packet Voice Protocols
Sponsored by:
Cisco Systems

Introduction
Over the past decade, the telecommunications industry has witnessed rapid changes in the way people and organizations communicate. Many of these changes spring from the explosive growth of the Internet and from applications based on the Internet Protocol (IP). The Internet has become a ubiquitous means of communication, and the total amount of packet-based network traffic has quickly surpassed traditional voice (circuit-switched) network traffic (DataQuest, 1998).

In the wake of these technology advancements, it has become clear to telecommunications carriers, companies, and vendors that voice traffic and services will be one of the next major applications to take full advantage of IP. This expectation is based on the impact of a new set of technologies generally referred to as voice over IP (VoIP) or IP telephony.

VoIP supplies many unique capabilities to the carriers and customers who depend on IP or other packet-based networks. The most important benefits include the following:

  • Cost Savings: By moving voice traffic to IP networks, companies can reduce or eliminate the toll charges associated with transporting calls over the public switched telephone network (PSTN). Service providers and end users can also conserve bandwidth by investing in additional capacity only when it is needed. This is made possible by the distributed nature of VoIP and by reduced operations costs as companies combine voice and data traffic onto one network.
  • Open Standards and Multivendor Interoperability: By adopting open standards, both businesses and service providers can purchase equipment from multiple vendors and eliminate their dependency on proprietary solutions.
  • Integrated Voice and Data Networks: By making voice "just another IP application," companies can build truly integrated networks for voice and data. These integrated networks not only provide the quality and reliability of today's PSTN, but they also enable companies to quickly and flexibly take advantage of new opportunities within the changing world of communications.

In 1995, the first commercial VoIP products began to hit the market. These products were targeted at companies looking to reduce telecommunications expenses by moving voice traffic to packet networks. Early adopters of VoIP networks built toll-bypass solutions to take advantage of favorable regulatory treatment of IP traffic. Without any established standards, most early implementations were based on proprietary technology.

As these packet telephony networks grew and interconnection dependencies emerged, it became clear that the industry needed standard VoIP protocols. Several groups took up the challenge, resulting in independent standards, each with its own unique characteristics. In particular, network equipment suppliers and their customers were left to sort out the similarities and differences between four different signaling and call-control protocols for VoIP:

  • H.323
  • Media Gateway Control Protocol (MGCP)
  • Session Initiation Protocol (SIP)
  • H.248/Media Gateway Control (MEGACO)

In the process of implementing workable VoIP solutions, network engineers had to determine how each of these protocols worked and which ones were best for particular networks and applications.

This paper provides some guidance and understanding of these VoIP protocols and tries to clarify some of the confusion in the marketplace.

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