Cisco Systems, 2008. — 310 pages.To provide voice communication over an IP network, Real-Time Transport Protocol (RTP) sessions are created. These sessions are dynamically created and facilitated by one of several call control procedures. Typically, these procedures also include mechanisms for signaling events during voice calls, and for managing and collecting statistics about the voice calls.
There are several types of protocols used within Cisco Unified Communications networks. In order to select the best protocol for a specific environment, network engineers need to know all of these protocols, how they can be used, and their advantages and disadvantages.
This module focuses on three protocols that are used in Cisco Unified Communications networks to implement gateways and offer call-control support for VoIP: the H.323 suite of protocols, the Media Gateway Control Protocol (MGCP), and the session initiation protocol (SIP).
