Implementing Multiple-Site Deployments
This chapter covers the following topics:
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Implementing multiple-site IP telephony deployments over an IP WAN requires additional planning to ensure the quality and availability of voice calls. When an IP WAN connects IP telephony clusters, a mechanism must exist to control the audio quality and video quality of calls over the IP WAN link by limiting the number of calls that are allowed on that link at the same time. Call admission control is the mechanism that ensures that voice calls do not oversubscribe the IP WAN bandwidth and affect voice quality.
When the priority queue of IP WAN bandwidth is consumed, a mechanism must exist to automatically reroute calls over the public switched telephone network (PSTN) without requiring the caller to hang up and redial the called party. Automated Alternate Routing (AAR) is a Cisco CallManager feature that automatically reroutes calls through the PSTN or other networks when priority bandwidth is insufficient in a centralized call-processing deployment.
Note
Referring to the priority queue of IP WAN bandwidth does not necessarily encompass the entire amount of WAN bandwidth available. Rather, it encompasses the amount of bandwidth you have set aside for high-priority traffic (including voice).
If connectivity with Cisco CallManager is lost, Cisco IP Phones become unusable for the duration of the failure. Cisco Survivable Remote Site Telephony (SRST) overcomes this problem and ensures that the Cisco IP Phones offer continuous service by providing call handling support for Cisco IP Phones directly from the Cisco SRST router.
This chapter describes the operation and configuration of call admission control, AAR, and SRST.