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An Overview of Cisco IP Communications

Chapter Description

This chapter provides an overview of Cisco IP Communications, including VoIP and how Cisco IP Communications differs from traditional telephone systems, and how you can use VoIP to achieve savings by routing your telephone calls over the IP WAN.


Like IP precedence, regions play an important role in ensuring the quality of voice calls within your network. Regions allow you to constrain the codecs selected when one device calls another. Most often, you use regions to limit the bandwidth used when calls are placed between devices connected by an IP WAN. However, you can also use regions as a way of providing higher voice quality at the expense of network bandwidth for a preferred class of users.

When you define a new region, Cisco CallManager Administration asks you to define the compression type used for calls between devices within the region. You also define, on a region-by-region basis, compression types used for calls between the region you are creating and all other regions.

You associate regions with device pools. All devices contained in a given device pool belong to the region associated with that device pool. When an endpoint in one device pool calls an endpoint in another, the codec used is constrained to what is defined in the region. If, for some reason, one of the endpoints in the call cannot encode the voice stream according to the specified codec, CallManager attempts to introduce a transcoder (see Chapter 5) to allow the endpoints to communicate.

Figure 1-24 depicts a configuration that uses three regions to constrain bandwidth between end devices. Phones 1000 and 2000 are in the main campus; phone 3000 is in a branch office. Calls within the main campus use the G.711 codec, as do calls from phone 1000 to phone 3000. Calls between phone 2000 and phone 3000 use the G.729 codec.


Figure 1-24 Regions Overview

CallManager Locations-Based Call Admissions Control

Locations represent a form of admissions control. A location defines a topological area connected to other areas by links of limited bandwidth. With each location, you specify the amount of bandwidth available between users in that location and other locations in your network.

CallManager allows users to place an unlimited number of calls between devices within the same location; when a user places a call to another location however, CallManager temporarily deducts the bandwidth associated with the selected codec from the interlocation bandwidth remaining. When a user's call terminates, CallManager returns the allocated bandwidth to the pool of available bandwidth.

Users who attempt to place a call when no more bandwidth is available receive a fast busy tone (also called reorder tone), unless you enable a feature called Automated Alternate Routing (AAR). If AAR is properly configured, then instead of rejecting calls when the bandwidth between locations is oversubscribed, if the dialed destination has a PSTN number, CallManager automatically redials it to send the call to the local branch gateway for routing the call over the public network.

You must consider several design caveats before using locations-based CAC.

  • Locations-based CAC requires that you deploy your voice network in a hub-and-spoke topology. Although locations allow you to configure admissions control, the locations mechanism is topologically ignorant. Having only one bandwidth counter for all interlocation calls means that all calls from one location to any other location must traverse only one logical network link, which limits deployment strictly to hub-and-spoke topologies. Figure 1-25 elaborates.

    Figure 1-25 Hub-and-Spoke Topology Restriction

  • When CallManager connects a call on behalf of a device that requires an MTP, CallManager does not account for the bandwidth between the device and the MTP. As a result, you must co-locate MTPs with the devices that require them and set up Media Resource Group Lists (MRGL) to use them.

H.323 Gatekeeper

CallManager can be configured to use an H.323 gatekeeper for call admissions control between CallManager clusters. Before placing an H.323 call, a gatekeeper-enabled CallManager makes a Registration, Admissions, and Status (RAS) protocol admissions request (ARQ) to the H.323 gatekeeper.

The H.323 gatekeeper associates the requesting CallManager with a zone and can track calls that come into and go out of the zone. If the bandwidth allocated for a particular zone is exceeded, the H.323 gatekeeper denies the call attempt, and the caller hears a fast busy tone. (Alternatively, using route lists, you can configure CallManager to offer the call to a local PSTN gateway if the gatekeeper denies the call.) Essentially, an H.323 gatekeeper provides a locations-like functionality for the H.323 domain. Figure 1-26 depicts a gatekeeper-enabled configuration.


Figure 1-26 H.323 Gatekeeper-Based Call Admissions Control

To configure fallback through the PSTN, you must configure the route plan to choose an alternate route if the gatekeeper rejects the call attempt. To configure PSTN fallback, you must configure a route list that contains two route groups. The first route group contains the intercluster trunk that routes outgoing calls over the IP WAN. If insufficient bandwidth is available, however, the H.323 gatekeeper rejects this outgoing call attempt. This call rejection triggers the alternate route associated with the route list. When CallManager selects this alternate route, it transforms the dialed digits to the destination's address as seen from the PSTN's point of view and offers the call to the PSTN gateway. Figure 1-26 demonstrates fallback routing through the PSTN.


Figure 1-27 Fallback Routing Through the PSTN

A call from Phone 1000 to Phone 5000 first attempts to route across the IP WAN. If the gatekeeper denies the call attempt, the route list modifies the dialed number and again offers the call to the gateway, which routes the call across the PSTN.

Chapter 2 discusses call routing in much more detail.