CCM Deployment Models

Wael K. Yousif @ Valencia Community College

Network Topologies
1. Single Site model 2. Multiple Site model with independent call processing 3. Multiple site IP WAN model with distributed call processing 4. Multiple site model with centralized call processing 5. Combined multiple site model
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Single-Site Model
No telephony services provided over an IP WAN. Single Cisco CallManager or Cisco CallManager cluster Maximum of 30,000 IP phones per cluster PSTN for all external calls Digital signal processor (DSP) resources for conferencing, transcoding, and media termination point (MTP) Voice mail and unified messaging components Only G.711 codecs for all IP phone calls (80 kbps of IP bandwidth per call, uncompressed) Capability to integrate with legacy private branch exchange (PBX) and voice mail systems
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Gateway Devices
Gateway devices provide access from one telephone system to another:
From one network of CallManager servers to another (H.323 trunks provide an alternative for connecting Call Manger network together without requiring a gateway device) From a CallManager network to a PBX From a CallManger network to a public network such as a Class 4 or Class 5 switch.
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Gateway Devices
CallManger supports:
H.323 gateway devices such as Cisco 2600 routers MGCP gateway devices such as Cisco catalyst 4000, and 6000 with voice Interface Cards Each Gateway type manages a set of traditional telephony interfaces.
Analog interfaces; same as the one runs into your home Digital interfaces; T1 Call Associate Signaling (CAS), or ISDN Primary Rate Interface

We will focus on Cisco 2600 H.323 gateway. (More on gateway devices later)
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Multi-Site WAN with Centralized Call Processing

The multi-site WAN model with centralized call processing consists of a single call processing agent that provides services for many sites and uses the IP WAN to transport IP telephony traffic between the sites. An IP WAN with QoS enabled (Priority Queuing, Traffic Shaping) to connect all the sites. The remote sites rely on the centralized Cisco CallManager cluster to handle their call processing. a call admission control scheme is needed to avoid oversubscribing the WAN links with voice traffic and deteriorating the quality of established calls.

The Survivable Remote Site Telephony (SRST) feature, available on Cisco IOS gateways, provides call processing at the branch offices in the event of a WAN failure.
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Call Admission Control for Centralized Call Processing

Multi-site deployments require some form of call admission control to ensure the voice quality of calls transmitted across network links that have limited available bandwidth. Cisco CallManager provides a simple mechanism known as locations for implementing call admission control in multi-site WAN deployments with centralized call processing. Follow these guidelines when using locations for call admission control: Locations require a hub-and-spoke network topology. Configure a separate location in Cisco CallManager for each site. Configure the appropriate bandwidth limit for each site according to the type of codec used at that site. Assign each device configured in Cisco CallManager to a location. If you move a device to another location, change its location configuration as well. Cisco CallManager supports up to 500 locations.

Automated Alternate Routing

The automated alternate routing (AAR) feature enables Cisco CallManager to establish an alternate path for the voice media when the preferred path between two intra-cluster endpoints runs out of available bandwidth, as determined by the locations mechanism for call admission control (CAC). The AAR feature applies primarily to centralized call processing deployments. For instance, if a phone in branch A calls a phone in branch B and the available bandwidth for the WAN link between the branches is insufficient (as computed by the locations mechanism), AAR can reroute the call through the PSTN. The audio path of the call would be IP-based from the calling phone to its local (branch A) PSTN gateway, TDM-based from that gateway through the PSTN to the branch B gateway, and IPbased from the branch B gateway to the destination IP phone. AAR can be transparent to the users. You can configure AAR so that users dial only the on-net (for example, 4-digit) directory number of the called phone and no additional user input is required to reach the destination 8 through the alternate network (such as the PSTN).

Multi-Site WAN with Distributed Call Processing

The multi-site WAN model with distributed call processing consists of multiple independent sites, each with its own call processing agent connected to an IP WAN that carries voice traffic between the distributed sites. Unlike the centralized call processing model, however, the IP WAN in the distributed model does not carry call control signaling between the sites because each site has its own call processing agent. Each site in the distributed call processing model can be one of the following: A single site with its own call processing agent, which can be either Cisco CallManager, Cisco IOS Telephony Services (ITS), or other IP PBX A centralized call processing site and all of its associated remote sites A legacy PBX with Voice over IP (VoIP) gateway
A gatekeeper is an H.323 device that provides call admission control and E.164 dial plan resolution.