Transcript Call Manager Basic Configuration

CCM Deployment Models
Wael K. Yousif @
Valencia Community College
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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.
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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 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
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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.
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