Cisco Unified CallManager

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Cisco Unified CallManager is the core call processing software for Cisco IP Telephony. It builds call processing capabilities on top of the Cisco IP network infrastructure. Cisco Unified CallManager extends enterprise telephony features and capabilities to telephony network devices such as IP phones, media processing devices, voice gateways, and multimedia applications.

CallManager can be deployed using one of the following models:

• Single Site: In this model each site or campus has its own CallManager and no voice traffic travels over the IP WAN instead they are just routed over the PSTN.
• Multi-site WAN with centralized call processing: In this scenario, CallManager cluster resides at the central campus and communication with the remote offices takes place over the IP WAN. It is recommended to use SRST (Survivable Remote Site Telephony) on remote sites in case of a loss of IP WAN.
• Multi-site WAN with distributed call processing: In this model, CallManagers are deployed at multiple locations and the VoIP traffic is transported over the IP WAN. Once again SRST should be utilized as a backup voice path.

Below are the best practices for each of the three models discussed above.

Best Practices of Single-site Model

• Most calls are within the same site or to PSTN users outside
• Highly available and fault-tolerant infrastructure with inline power, QoS and security for IP phones
• Use the Media Gateway Control Protocol (MGCP) gateways for the PSTN if H.323 functionality is not required.
• Use G.711 codecs for all endpoints so you won't consume the DSP (digital signal processor) resources for transcoding.

Best Practices for Centralized Call Processing Model

• Minimize delay between Cisco Unified CallManager and remote sites
• Consider various factors like WAN Bandwidth, delay, scalability, cost, ease of management etc.
• Use Locations mechanism  to provide Call Admission Control (CAC) into and out of remote offices.
• Deploy Survivable Remote Site Telephony (SRST) where possible.

Best Practices for Distributed Call Processing Model

• Follow the best practices for Single-site and Centralized Call Processing models
• Use a Cisco IOS gatekeeper to provide CAC into and out of each site and SIP proxies for dial-plan resolution.
• Use HSRP gatekeeper pairs, gatekeeper clustering and alternate gatekeeper support for high availability. SIP proxies should also have redundancy enabled.
• Use only one type of codec on the WAN.

IP Telephony migration options can be categorized into the following categories:

Phased Migration:

This migration starts off as a limited initial IP Telephony deployment that is connected to the main corporate PBX. A gateway is used to provide connecticity between CallManager and PBX.

Cisco CallManager supports either regular PSTN-type PRI or QSIG PRI along with H.323 and SIP. T1/E1 QSIG is able to provide the greatest level feature transparency between the two systems.

The migration spans over several phases where users are moved from PBX to Cisco Unified CallManager in groups until everyone is migrated over.

Parallel Migration:

This approach starts off with an implementation of a complete, parallel IP Telephony infrastructure that is redundant and highly available. End users have (2) IP Phones on their desk simultaneously providing opportunity to test and train on the system.

Once the IP Telephony system is fully deployed, a change window can be scheduled to decommission  the PBX system.