Two Voice Implementations

Voice transport is a general term that can be divided into the following two implementations:

■ VoIP: VoIP uses voice-enabled routers to convert analog voice into IP packets or packetized digital voice channels and route those packets between corresponding locations. Users do not often notice that VoIP is implemented in the network—they use their traditional phones, which are connected to a PBX. However, the PBX is not connected to the PSTN or to another PBX, but to a voice-enabled router that is an entry point to VoIP. Voice-enabled routers can also terminate IP phones using Session Initiation Protocol for call control and signaling.

■ IP telephony: For IP telephony, traditional phones are replaced with IP phones. A server for call control and signaling, such as a Cisco Unified Communications Manager, is also used. The IP phone itself performs voice-to-IP conversion, and no voice-enabled routers are required within the enterprise network. However, if a connection to the PSTN is required, a voice-enabled router or other gateway in the Enterprise Edge is added where calls are forwarded to the PSTN.

NOTE Earlier names for the Cisco Unified Communications Manager include Cisco CallManager and Cisco Unified CallManager.

Both implementations require properly designed networks. Using a modular approach in a voice transport design is especially important because of the voice sensitivity to delay and the complexity of troubleshooting voice networks. All Cisco Enterprise Architecture modules are involved in voice transport design.

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