CCM Deployment Models

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CCM Deployment Models

• Wael K. Yousif _at_
• Valencia Community College

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Network Topologies

• Single Site model
• Multiple Site model with independent call
  processing
• Multiple site IP WAN model with distributed call
  processing
• Multiple site model with centralized call
  processing
• 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.

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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 IP-based 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 through the
  alternate network (such as the PSTN).

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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.