IP telephony can be designed in a network in many scenarios, including single-site and multisite deployments. Single-Site IP Telephony DesignA single-site design has all the IP telephony components at one site, as illustrated in Figure 7-3. Figure 7-3. IP Telephony Single-Site Design
In this scenario, the IP phones (connected to switches that can provide them in-line power), a call-processing engine (CCM), application servers, and optionally a voice gateway to the PSTN are at the same physical location. Each site is self-contained, and all calls between sites are through the PSTN. This means that the IP WAN is not involved in voice calls and is therefore not a voice bottleneck. Multisite Centralized IP Telephony DesignA multisite design has many sites, interconnected through a WAN. A centralized design means that the call-processing engine and application servers are at one of the sites, and the other sites connect to those devices for all call-processing and application requirements, as illustrated in Figure 7-4. Figure 7-4. IP Telephony Multisite Centralized Design
Because the remote sites must send call-processing information to the main site, they must have IP connectivity with the main site. Therefore, if the WAN is down, calls cannot be processed. To prevent a total breakdown of the IP telephony system, the Survivable Remote Site Telephony (SRST) feature on remote routers can be used to provide basic call processing to the remote sites if the WAN fails. If PSTN access is not required, a voice-enabled router is not necessary in this scenario, because only IP packets are sent across the WAN. However, if PSTN access is required (which is the usual case), a voice gateway is required. Multisite Distributed IP Telephony DesignSimilar to the centralized design, a distributed multisite network has many sites, interconnected through a WAN. In this case, though, each site has call-processing resources. Calls between sites can be through the IP WAN or through the PSTN. |