VoIP and Unified Communications: Internet Telephony and the Future Voice Network

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VoIP SIGNALING AND CALL PROCESSING

A fundamental assumption underlies the basic standards and recommendations for signaling in the IP environment: every device attaches directly to the open Internet and has a public, routable (i.e., easily reachable) address. Transactions to set up, modify, and end sessions—as defined in their primary documents—generally don’t consider firewalls, network address translation, access controls, or other security measures that might exist at the boundary between the Internet and an enterprise LAN. Many of those security measures prevent VoIP signaling and conversation flows. Workarounds allow VoIP to work in practice on today’s Internet, using IP version 4.

Examples abound of protocols that assume open connectivity: the basic RFCs for signaling protocols, the SIP Forum recommendation for SIP trunking, and most other descriptions of IP telephony. This section likewise will treat signaling first in isolation, to describe its functioning. A later section will describe how to communicate through the barriers set up by security measures.

There are three primary signal formats for VoIP:

• The oldest is H.323, a family of protocols created by the ITU with a goal of enabling the functions needed by carriers to provide not only a service similar to that of the PSTN but also one- and two-way video distribution.
• Session Initiation Protocol (SIP) is the latest version of signaling and has dominated development work since about 2000. All new products support it, ...