Campus Clustering
As we have discussed previously, it is assumed there is fast, LAN connectivity
available between servers within a campus.This is essential because with Cisco
CallManager 3.x, clustering across a WAN is not supported. For most campusbased
networks, a single cluster solution is adequate. Call admission control is not
required within a campus, but the following restrictions do apply:
Maximum of 10,000 total registered devices
Maximum of eight servers per cluster
Maximum of 2,500 registered IP phones, or 3,000 other devices per
CallManager
Switched infrastructure to the desktop
You should ensure that the maximum redundancy and load-balancing options
are provided by your cluster. It is important to consider that typically, some sites
within a campus will have only a single high-speed IP link to the rest of the
MAN. In such cases, it is essential that CallManagers are placed on site, to ensure
system availability in the event of a link failure. Figure 4.5 illustrates a typical
MAN, and how clustering might be used to ensure a high-availability and highperformance
system.
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Table 4.4 Continued
Max Device Max IP Phones
Server Platform Specification Units per Server per Server
IT Certification CCIE,CCNP,CCIP,CCNA,CCSP,Cisco Network Optimization and Security Tips
Maximum Device Units per Server Platform
Maximum Device Units per Server Platform
Max Device Max IP Phones
Server Platform Specification Units per Server per Server
MCS-7835-1000 5,000 2,500
PIII 1000MHz, 1GB RAM
MCS-7835 5,000 2,500
PIII 733MHz, 1GB RAM
MCS-7830 3,000 1,500
PIII 500MHz, 1GB RAM
MCS-7830 1,000 500
PIII 500MHz, 512MB RAM
MCS-7825-800 1,000 500
PIII 800MHz, 512MB RAM
MCS-7822 1,000 500
PIII 550MHz, 512MB RAM
MCS-7820 1,000 500
Max Device Max IP Phones
Server Platform Specification Units per Server per Server
MCS-7835-1000 5,000 2,500
PIII 1000MHz, 1GB RAM
MCS-7835 5,000 2,500
PIII 733MHz, 1GB RAM
MCS-7830 3,000 1,500
PIII 500MHz, 1GB RAM
MCS-7830 1,000 500
PIII 500MHz, 512MB RAM
MCS-7825-800 1,000 500
PIII 800MHz, 512MB RAM
MCS-7822 1,000 500
PIII 550MHz, 512MB RAM
MCS-7820 1,000 500
Weight Chart for IP Telephony Devices
Weight Chart for IP Telephony Devices
Weight per Instance per Total Device
Device Type Instance Device Weight
IP phone 1 1 1
Analog gateway ports 3 Variable 3 per DS0
T1 gateway 3 24 72 per T1
E1 gateway 3 30 90 per E1
Conference resource 3 Variable 3 per instance
(hardware)
Conference resource 3 48 144*
(software)
Transcoding resource 3 Variable 3 per instance
Software MTP 3 48 144*
CTI port (TAPI or JTAPI) 20 1 20
Messaging (voice mail) 3 Variable 3 per instance
Cisco SoftPhone 20 1 20
Inter-cluster trunk 3 Variable 3
*If installed on the same server as Cisco CallManager, the maximum number
of sessions is 48.
For example, an infrastructure containing 1,250 IP phones, 2 T1 gateways, a
software conference resource, and two Cisco SoftPhones, would constitute the
following formula: (1,250 x 1) + (2 x 72) + (144 x 1) + (20 x 2) = 1,540 total
device weight (device units).Table 4.4 details the maximum number of device
units that may be serviced by specific server platforms. Currently, you cannot
have more than 2,500 IP phones registered with a single Cisco CallManager,
even if the maximum device units allow it.
Weight per Instance per Total Device
Device Type Instance Device Weight
IP phone 1 1 1
Analog gateway ports 3 Variable 3 per DS0
T1 gateway 3 24 72 per T1
E1 gateway 3 30 90 per E1
Conference resource 3 Variable 3 per instance
(hardware)
Conference resource 3 48 144*
(software)
Transcoding resource 3 Variable 3 per instance
Software MTP 3 48 144*
CTI port (TAPI or JTAPI) 20 1 20
Messaging (voice mail) 3 Variable 3 per instance
Cisco SoftPhone 20 1 20
Inter-cluster trunk 3 Variable 3
*If installed on the same server as Cisco CallManager, the maximum number
of sessions is 48.
For example, an infrastructure containing 1,250 IP phones, 2 T1 gateways, a
software conference resource, and two Cisco SoftPhones, would constitute the
following formula: (1,250 x 1) + (2 x 72) + (144 x 1) + (20 x 2) = 1,540 total
device weight (device units).Table 4.4 details the maximum number of device
units that may be serviced by specific server platforms. Currently, you cannot
have more than 2,500 IP phones registered with a single Cisco CallManager,
even if the maximum device units allow it.
Device Weights
Device Weights
Each IP telephony device that registers with CiscoManager is assigned a weight.
The weight assigned to devices, such as IP phones, H.323 gateways, conferencing
hardware,Telephony Application Programming Interfaces (TAPI), and Java TAPI
(JTAPI), is based on the memory and CPU resources that they consume.The
higher the weight allocated, the more resources they consume.
By considering the weights of devices to be registered in your network, you
can work out the number of CallManagers required, and the optimal hardware
specification of that hardware.Table 4.3 shows the weights allocated to each
device type.
Each IP telephony device that registers with CiscoManager is assigned a weight.
The weight assigned to devices, such as IP phones, H.323 gateways, conferencing
hardware,Telephony Application Programming Interfaces (TAPI), and Java TAPI
(JTAPI), is based on the memory and CPU resources that they consume.The
higher the weight allocated, the more resources they consume.
By considering the weights of devices to be registered in your network, you
can work out the number of CallManagers required, and the optimal hardware
specification of that hardware.Table 4.3 shows the weights allocated to each
device type.
ip phone-cisco
5,000 Four CallManager servers: Two groups, servers BD,
Database publisher/TFTP and CD:
server (A) Server B primary CM for
Two primary Cisco IP phones 0–2500
CallManagers (B and C) Server C primary CM for
Backup Cisco IP phones 2501–5000
CallManager (D) Server D secondary CM for
all IP phones
10,000 Eight CallManager servers: Four groups, servers CE, DE,
Database publisher (A) FH, and GH:
TFTP server (B) Server C primary CM for IP
Four primary Cisco phones 0–2500
CallManagers (C, D, E, and F) Server D primary CM for IP
Two backup Cisco phones 2501–5000
CallManagers (G and H) Server E backup CM for IP
phones 1–5000
Server F primary CM for IP
phones 5001–7500
Server G primary CM for IP
phones 7501–10000
Server H backup CM for IP
phones 5001–10000
Database publisher/TFTP and CD:
server (A) Server B primary CM for
Two primary Cisco IP phones 0–2500
CallManagers (B and C) Server C primary CM for
Backup Cisco IP phones 2501–5000
CallManager (D) Server D secondary CM for
all IP phones
10,000 Eight CallManager servers: Four groups, servers CE, DE,
Database publisher (A) FH, and GH:
TFTP server (B) Server C primary CM for IP
Four primary Cisco phones 0–2500
CallManagers (C, D, E, and F) Server D primary CM for IP
Two backup Cisco phones 2501–5000
CallManagers (G and H) Server E backup CM for IP
phones 1–5000
Server F primary CM for IP
phones 5001–7500
Server G primary CM for IP
phones 7501–10000
Server H backup CM for IP
phones 5001–10000
Balanced Call Processing
Balanced Call Processing
Server A
Server B
Server C
Primary
Secondary
Tertiary
Server B
Server A
Server C
Primary
Tertiary
Secondary
Server Group 1 Server Group 2
Device Group 1
Device Group 2
Server A
Server B
Server C
Primary
Secondary
Tertiary
Server B
Server A
Server C
Primary
Tertiary
Secondary
Server Group 1 Server Group 2
Device Group 1
Device Group 2
Recommended CallManager Configurations
Recommended CallManager Configurations
Recommended CallManager
IP Phones Configuration Redundancy Groups
2,500 Three CallManager servers: One group, servers AB:
Database publisher/TFTP Server A primary CM for
server (A) all IP phones
Primary Cisco Server B backup CN for all
CallManager (B) IP phones
Backup Cisco
CallManager (C)
Recommended CallManager
IP Phones Configuration Redundancy Groups
2,500 Three CallManager servers: One group, servers AB:
Database publisher/TFTP Server A primary CM for
server (A) all IP phones
Primary Cisco Server B backup CN for all
CallManager (B) IP phones
Backup Cisco
CallManager (C)
Designing CallManager Clusters
Designing CallManager Clusters
The powerful features offered by a CallManager cluster depend heavily on a good
design methodology. It is very important for you to have a good understanding of
the limitations, and Cisco recommendations, of CallManager clustering, in order to
design a quality IP telephony infrastructure.With this in place, you will be able to
provide a robust, efficient, and feature-rich IP telephony solution.
With Cisco CallManager 3.0(5) we are allowed a maximum of eight servers
per cluster, of which a maximum of six may be used for call processing.The
remaining two servers should be used as a database publisher server, and TFTP
server.These two roles may be combined on the same server, but it is recommended
they reside on separate servers for large installations.There can only be
one database publisher server, and one TFTP server per cluster.
Cisco recommends the guidelines shown in Table 4.2 when deciding upon the
configuration of a CallManager cluster. It is possible to use fewer servers, but
redundancy and load balancing will be reduced.The table also indicates the recommended
redundancy group configuration.A maximum of 2,500 IP phones, gateways,
and DSPs can be allocated to each CallManager with a maximum of 5,000
devices in total.
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The powerful features offered by a CallManager cluster depend heavily on a good
design methodology. It is very important for you to have a good understanding of
the limitations, and Cisco recommendations, of CallManager clustering, in order to
design a quality IP telephony infrastructure.With this in place, you will be able to
provide a robust, efficient, and feature-rich IP telephony solution.
With Cisco CallManager 3.0(5) we are allowed a maximum of eight servers
per cluster, of which a maximum of six may be used for call processing.The
remaining two servers should be used as a database publisher server, and TFTP
server.These two roles may be combined on the same server, but it is recommended
they reside on separate servers for large installations.There can only be
one database publisher server, and one TFTP server per cluster.
Cisco recommends the guidelines shown in Table 4.2 when deciding upon the
configuration of a CallManager cluster. It is possible to use fewer servers, but
redundancy and load balancing will be reduced.The table also indicates the recommended
redundancy group configuration.A maximum of 2,500 IP phones, gateways,
and DSPs can be allocated to each CallManager with a maximum of 5,000
devices in total.
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Balanced Call Processing
Balanced Call Processing
You can balance calls between the servers in the CallManager cluster through the
use of CallManager groups and device pools, as defined in the previous section.
By allocating different devices to different pools, and then assigning these pools
to different CallManager groups, you have a very flexible method of achieving
the load balancing you require.
It is recommended there be some common characteristic shared by each
device in a pool, such as physical location, IP subnet, or device type.
In Figure 4.4, we can see two device groups, and two server groups.Within
our server group we have three servers; Server A and B are dedicated call processing
servers, and Server C is configured as a combined database publisher and
TFTP server, but also has call processing capabilities. Server group 1 defines
Server A as the primary server, Server B as the secondary server, and Server C as
the tertiary server. Server group 2 defines Server B as the primary server, Server
A as the secondary server, and Server C as the tertiary server. Device group 1
uses Server group 2, and Device group 2 uses Server group 1.This means that
under normal circumstances call processing would be evenly balanced between
Server A and B. Server C would only be used for call processing functions if both
Server A and B became unavailable.
You can balance calls between the servers in the CallManager cluster through the
use of CallManager groups and device pools, as defined in the previous section.
By allocating different devices to different pools, and then assigning these pools
to different CallManager groups, you have a very flexible method of achieving
the load balancing you require.
It is recommended there be some common characteristic shared by each
device in a pool, such as physical location, IP subnet, or device type.
In Figure 4.4, we can see two device groups, and two server groups.Within
our server group we have three servers; Server A and B are dedicated call processing
servers, and Server C is configured as a combined database publisher and
TFTP server, but also has call processing capabilities. Server group 1 defines
Server A as the primary server, Server B as the secondary server, and Server C as
the tertiary server. Server group 2 defines Server B as the primary server, Server
A as the secondary server, and Server C as the tertiary server. Device group 1
uses Server group 2, and Device group 2 uses Server group 1.This means that
under normal circumstances call processing would be evenly balanced between
Server A and B. Server C would only be used for call processing functions if both
Server A and B became unavailable.
Cisco CallManager Redundancy
Cisco CallManager Redundancy
Device Pool
Primary
Secondary
Tertiary
IP telephones, and other IP telephony devices such as gateways, transcoders,
conference bridges, and voice mail ports can be assigned to a device pool.A device
pool is defined as a group of IP telephony devices that share the same characteristics.
These characteristics are defined when the device pool is created, and include
name, region, media resource group list, user hold Music On Hold source, autoanswer
feature control, and most importantly the associated Cisco CallManager
group.As with CallManager groups, each device within a pool should share a similar
characteristic, such as geographical location, or logical subnet.
IP phones within a pool maintain a TCP connection with both the primary
and secondary CallManager servers in the CallManager group associated with
that pool.This facilitates immediate failover should the primary CallManager
server fail.
Device Pool
Primary
Secondary
Tertiary
IP telephones, and other IP telephony devices such as gateways, transcoders,
conference bridges, and voice mail ports can be assigned to a device pool.A device
pool is defined as a group of IP telephony devices that share the same characteristics.
These characteristics are defined when the device pool is created, and include
name, region, media resource group list, user hold Music On Hold source, autoanswer
feature control, and most importantly the associated Cisco CallManager
group.As with CallManager groups, each device within a pool should share a similar
characteristic, such as geographical location, or logical subnet.
IP phones within a pool maintain a TCP connection with both the primary
and secondary CallManager servers in the CallManager group associated with
that pool.This facilitates immediate failover should the primary CallManager
server fail.
Redundancy within a CallManager Cluster
Redundancy within a CallManager Cluster
There are two types of redundancy within a CallManager Cluster: database redundancy
and server failover. Database redundancy is achieved by the replication of the
publisher database using the Publisher/Subscriber relationship.This ensures that
the same database is held by all servers within a cluster, which means the database
is accessible even if a server is not operational. Server failover occurs when a
CallManager server being used by an IP Telephony device fails.The device then
uses a predefined alternative server to carry out the required tasks.When the preferred
server becomes available again, the device switches back to using it.
Redundancy is achieved by configuring CallManager redundancy groups, and
then assigning pools of devices to use these groups. A CallManager redundancy
group consists of up to three servers—a primary, secondary, and tertiary. If the
primary server fails, the secondary server is used by a device in an associated
pool, and if the secondary fails, then the tertiary is used. If at any time the primary
server becomes operational again, the devices revert back to using that
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AVVID Clustering • Chapter 4 107
server. All members of a redundancy group must also be a member of the same
cluster. Figure 4.3 illustrates the relationship between devices in a device pool,
and CallManager servers within a redundancy group.
There are two types of redundancy within a CallManager Cluster: database redundancy
and server failover. Database redundancy is achieved by the replication of the
publisher database using the Publisher/Subscriber relationship.This ensures that
the same database is held by all servers within a cluster, which means the database
is accessible even if a server is not operational. Server failover occurs when a
CallManager server being used by an IP Telephony device fails.The device then
uses a predefined alternative server to carry out the required tasks.When the preferred
server becomes available again, the device switches back to using it.
Redundancy is achieved by configuring CallManager redundancy groups, and
then assigning pools of devices to use these groups. A CallManager redundancy
group consists of up to three servers—a primary, secondary, and tertiary. If the
primary server fails, the secondary server is used by a device in an associated
pool, and if the secondary fails, then the tertiary is used. If at any time the primary
server becomes operational again, the devices revert back to using that
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AVVID Clustering • Chapter 4 107
server. All members of a redundancy group must also be a member of the same
cluster. Figure 4.3 illustrates the relationship between devices in a device pool,
and CallManager servers within a redundancy group.
Multisite WANs
Multisite WANs with centralized call processing (No CallManager
at remote sites) The Cisco CallManager locations feature is used to
implement call admission control.
Unfortunately, most Cisco CallManager features do not function between
CallManager clusters.You must therefore consider that only the following features
will exist between clusters:
Basic call setup
G.711 and G.729 calls
Call transfer
Call park
Call hold
Calling line ID
Multiparty conference
For this reason, you must take care when designing your IP Telephony network,
in order to reduce unnecessary CallManager clusters.
at remote sites) The Cisco CallManager locations feature is used to
implement call admission control.
Unfortunately, most Cisco CallManager features do not function between
CallManager clusters.You must therefore consider that only the following features
will exist between clusters:
Basic call setup
G.711 and G.729 calls
Call transfer
Call park
Call hold
Calling line ID
Multiparty conference
For this reason, you must take care when designing your IP Telephony network,
in order to reduce unnecessary CallManager clusters.
Multisite WANs
Multisite WANs with distributed call processing (At least one
CallManager at remote sites) Gatekeepers are used to provide call
admission control.
CallManager at remote sites) Gatekeepers are used to provide call
admission control.
Large MANs
Large MANs These are expected to have ample bandwidth to carry
the CallManager traffic, and therefore require no call admission control.
the CallManager traffic, and therefore require no call admission control.
Inter-Cluster Communication
Inter-Cluster Communication
Inter-cluster communication is often required in very large, or widely distributed
installations, where more than one CallManager cluster is required to handle the
load. H.323 inter-cluster trunks are established to communicate between the different
clusters.
The latest release of Cisco CallManager can support up to eight servers
within a cluster, and all members of the cluster must reside on the same LAN.
However, even though an individual CallManager server can support up to 2,500
IP phones, there is a fixed limit of 10,000 IP phones per cluster. From these limitations,
it is obvious that several clusters would be required in very large local
networks, or any geographically dispersed network.
There are three types of implementations that might require multiple clusters
as well as inter-cluster communication:
Inter-cluster communication is often required in very large, or widely distributed
installations, where more than one CallManager cluster is required to handle the
load. H.323 inter-cluster trunks are established to communicate between the different
clusters.
The latest release of Cisco CallManager can support up to eight servers
within a cluster, and all members of the cluster must reside on the same LAN.
However, even though an individual CallManager server can support up to 2,500
IP phones, there is a fixed limit of 10,000 IP phones per cluster. From these limitations,
it is obvious that several clusters would be required in very large local
networks, or any geographically dispersed network.
There are three types of implementations that might require multiple clusters
as well as inter-cluster communication:
Intra-Cluster Database Configuration
Intra-Cluster Database Configuration
Publisher Database (SQL)
Subscriber
Subscriber Subscriber
Subscriber
AVVID Clustering • Chapter 4 105
The real-time data that is communicated by servers within the cluster consists
of information such as the registration of IP phones, H.323 gateways, and digital
signal processors (DSPs).This information is replicated using a full-mesh, peer-topeer
model (see Figure 4.2) to ensure consistent and optimum routing of calls
throughout the system.
Publisher Database (SQL)
Subscriber
Subscriber Subscriber
Subscriber
AVVID Clustering • Chapter 4 105
The real-time data that is communicated by servers within the cluster consists
of information such as the registration of IP phones, H.323 gateways, and digital
signal processors (DSPs).This information is replicated using a full-mesh, peer-topeer
model (see Figure 4.2) to ensure consistent and optimum routing of calls
throughout the system.
Feature Transparency CallManager
Feature Transparency CallManager clustering provides the transparent
support of user features across a high-speed campus or metropolitan area
networks (MANs).
support of user features across a high-speed campus or metropolitan area
networks (MANs).
Intra-Cluster Communication
Intra-Cluster Communication
As the name suggests, intra-cluster communication is the communication between
servers in the same cluster.Two types of information are communicated between
servers in the same cluster—CallManager database information, and real-time data.
The CallManager database contains the configuration of all IP telephony
devices.When this configuration is updated in the CallManager Administrator,
the information is stored in the local database of the CallManager Publisher.The
Database Publisher then sends this information to all the database subscribers in
the cluster, who then update their local copies of the database.
As you can see in Figure 4.1, CallManager database replication works as a
client/server model.This feature allows for consistency between databases, and
redundancy in the event of a server failure.
As the name suggests, intra-cluster communication is the communication between
servers in the same cluster.Two types of information are communicated between
servers in the same cluster—CallManager database information, and real-time data.
The CallManager database contains the configuration of all IP telephony
devices.When this configuration is updated in the CallManager Administrator,
the information is stored in the local database of the CallManager Publisher.The
Database Publisher then sends this information to all the database subscribers in
the cluster, who then update their local copies of the database.
As you can see in Figure 4.1, CallManager database replication works as a
client/server model.This feature allows for consistency between databases, and
redundancy in the event of a server failure.
CallManager Cluster Communications
CallManager Cluster Communications
There are two types of communication between CallManager clusters:
Intra-cluster communication
Inter-cluster communication
Intra-cluster communication is defined as communications between
CallManager servers in the same cluster. It consists of replication of the
CallManager Database as well as dynamic information such as the registration of
H.323 devices. Inter-cluster communication deals with communications between
different clusters, and is established through the use of inter-cluster trunks.
There are two types of communication between CallManager clusters:
Intra-cluster communication
Inter-cluster communication
Intra-cluster communication is defined as communications between
CallManager servers in the same cluster. It consists of replication of the
CallManager Database as well as dynamic information such as the registration of
H.323 devices. Inter-cluster communication deals with communications between
different clusters, and is established through the use of inter-cluster trunks.
Scalability By using several CallManager
Scalability By using several CallManager servers with a common database,
call processing and other functions can be distributed throughout
the cluster.This will ease the load on individual CallManager servers,
and allow for localized processing of calls. Using this feature can help
facilitate efficient call processing over large wide area networks (WANs).
call processing and other functions can be distributed throughout
the cluster.This will ease the load on individual CallManager servers,
and allow for localized processing of calls. Using this feature can help
facilitate efficient call processing over large wide area networks (WANs).
Resiliency/Redundancy/Survivability
Resiliency/Redundancy/Survivability If a CallManager server in a
cluster fails, a different operational server can be used by the client for
call-processing functions.The publisher database containing IP device
configuration information is replicated throughout the cluster; in the
event of a server failure the database is not lost
cluster fails, a different operational server can be used by the client for
call-processing functions.The publisher database containing IP device
configuration information is replicated throughout the cluster; in the
event of a server failure the database is not lost
CallManager Roles within a Cluster
CallManager Roles within a Cluster
CallManager Role Description
Database publisher server Makes all configuration changes, and produces
call detail records.
TFTP server Handles downloading of configuration files,
ring types, and device operating code.
Application software server Software installed adds features to the IP
telephony solution.
Primary call-processing server Responsible for call-processing functions.
Backup call-processing server Responsible for call-processing functions.
Each CallManager in the server may be assigned one or more of these roles,
but there is only one database publisher server, and one Trivial File Transfer
Protocol (TFTP) server per cluster.You must decide on the level of redundancy
and processing distribution required in your installation. For larger installations
it is recommended to split the database publisher server and TFTP servers onto
different servers.
CallManager Role Description
Database publisher server Makes all configuration changes, and produces
call detail records.
TFTP server Handles downloading of configuration files,
ring types, and device operating code.
Application software server Software installed adds features to the IP
telephony solution.
Primary call-processing server Responsible for call-processing functions.
Backup call-processing server Responsible for call-processing functions.
Each CallManager in the server may be assigned one or more of these roles,
but there is only one database publisher server, and one Trivial File Transfer
Protocol (TFTP) server per cluster.You must decide on the level of redundancy
and processing distribution required in your installation. For larger installations
it is recommended to split the database publisher server and TFTP servers onto
different servers.
CallManager Clustering
CallManager Clustering
CallManager clustering is a method of seamlessly distributing call processing
throughout a converged IP network. By using clustering, several CallManager
servers can share the burden of call processing, which becomes particularly
important in larger or widely distributed IP Telephony implementations.
A cluster is defined as a set of Cisco CallManager servers sharing the same
database and resources.The roles described in Table 4.1 can be assigned to members
of a CallManager cluster.
CallManager clustering is a method of seamlessly distributing call processing
throughout a converged IP network. By using clustering, several CallManager
servers can share the burden of call processing, which becomes particularly
important in larger or widely distributed IP Telephony implementations.
A cluster is defined as a set of Cisco CallManager servers sharing the same
database and resources.The roles described in Table 4.1 can be assigned to members
of a CallManager cluster.
AVVID-CallManager
The AVVID solution provides options that allow scaling and load-distribution for
both IP telephony and voice/video conferencing features. Clustering is a technique
used to enhance both the capabilities of the network as well as the redundancy
within the network.With the networking capabilities available today, it is
possible to cluster both voice networks and video networks. Clustering techniques
allow you to scale your networks and, when the need arises, to add more
users or services.
Cisco CallManager clusters can be configured to distribute call processing
and device registration between multiple servers on the same segment. Up to
eight servers can be part of a cluster, with common database information and
real-time device registration data being replicated throughout using intra-cluster
communications.With common information shared throughout a number of
servers, redundancy is achieved. If a single server fails, another server can transparently
take over call processing for a group of devices. For very large or multisite
installations, several clusters may be used, with information being shared via intercluster
communication. However, you should use this solution with care, as many
Cisco CallManager features will not function between clusters.
Voice and video conferencing is facilitated by terminals producing
voice/video data streams, and by Multipoint Control Units, which control the
conference. For voice or video conferences larger than those supported by a
single MCU, Cisco offers a feature known as cascading. Cascading allows you to
cascade two or more MCUs, in order to provide a single larger conference. In
addition to providing highly scalable conferences, this also provides load distribution
between multiple MCUs, and allows voice and video streams to be localized
by the use of MCUs on different segments.
both IP telephony and voice/video conferencing features. Clustering is a technique
used to enhance both the capabilities of the network as well as the redundancy
within the network.With the networking capabilities available today, it is
possible to cluster both voice networks and video networks. Clustering techniques
allow you to scale your networks and, when the need arises, to add more
users or services.
Cisco CallManager clusters can be configured to distribute call processing
and device registration between multiple servers on the same segment. Up to
eight servers can be part of a cluster, with common database information and
real-time device registration data being replicated throughout using intra-cluster
communications.With common information shared throughout a number of
servers, redundancy is achieved. If a single server fails, another server can transparently
take over call processing for a group of devices. For very large or multisite
installations, several clusters may be used, with information being shared via intercluster
communication. However, you should use this solution with care, as many
Cisco CallManager features will not function between clusters.
Voice and video conferencing is facilitated by terminals producing
voice/video data streams, and by Multipoint Control Units, which control the
conference. For voice or video conferences larger than those supported by a
single MCU, Cisco offers a feature known as cascading. Cascading allows you to
cascade two or more MCUs, in order to provide a single larger conference. In
addition to providing highly scalable conferences, this also provides load distribution
between multiple MCUs, and allows voice and video streams to be localized
by the use of MCUs on different segments.
Multimedia Conference Manager Services
chp 4
Multimedia Conference Manager Services
.. Multimedia Conference Manager (MCM) works in conjunction with
Cisco’s IP/VC products, and services a H.323 gatekeeper and proxy.
.. MCM is a part of the Cisco IOS for the following router platforms:
2500, 2600, 3600, 3810, and 7200.
.. The MCM gatekeeper functions include: zone administration, RAS,
AAA services, bandwidth management, session management, and call
accounting.The proxy service provides QoS capabilities to the
videoconferencing sessions.
Multimedia Conference Manager Services
.. Multimedia Conference Manager (MCM) works in conjunction with
Cisco’s IP/VC products, and services a H.323 gatekeeper and proxy.
.. MCM is a part of the Cisco IOS for the following router platforms:
2500, 2600, 3600, 3810, and 7200.
.. The MCM gatekeeper functions include: zone administration, RAS,
AAA services, bandwidth management, session management, and call
accounting.The proxy service provides QoS capabilities to the
videoconferencing sessions.
Choosing a Voice Gateway Solution
.. Determining the right voice gateway solutions will depend on a number
of factors, from the size and scale of the organization to the budget.
.. Solutions from a switch point-of-view would include, the Catalyst 4000,
4224/4248, and 6000 family. If you wish to use routers, you should
choose from the following: the 1750, 2600, 3600, 3810, 7200, and 7500
Series. Access servers may be best in some instances, including the
AS5300, the AS5400, and the AS5800. Cisco DT-24, DE-30, and VG-
200 would suffice for standalone protocol solutions.
.. For small- to mid-sized companies looking for a nice all-in-one
solution, the ICS 7750, deployed with a Catalyst 3524XL-PWR switch
and Cisco IP phones, would do wonderfully.
.. The DPA 7610/7630 Voice Mail Gateway would be another important
element of an AVVID solution. It provides a gateway allowing legacy
voice mail systems to communicate with Cisco CallManagers.
.. Determining the right voice gateway solutions will depend on a number
of factors, from the size and scale of the organization to the budget.
.. Solutions from a switch point-of-view would include, the Catalyst 4000,
4224/4248, and 6000 family. If you wish to use routers, you should
choose from the following: the 1750, 2600, 3600, 3810, 7200, and 7500
Series. Access servers may be best in some instances, including the
AS5300, the AS5400, and the AS5800. Cisco DT-24, DE-30, and VG-
200 would suffice for standalone protocol solutions.
.. For small- to mid-sized companies looking for a nice all-in-one
solution, the ICS 7750, deployed with a Catalyst 3524XL-PWR switch
and Cisco IP phones, would do wonderfully.
.. The DPA 7610/7630 Voice Mail Gateway would be another important
element of an AVVID solution. It provides a gateway allowing legacy
voice mail systems to communicate with Cisco CallManagers.
Solutions Fast Track-Introduction to AVVID Gateways
Solutions Fast Track
Introduction to AVVID Gateways
.. In the Cisco AVVID world, there are voice and video gateways to provide
connectivity to legacy networks. Cisco has voice gateways, which are
standalone routers, IOS-based routers, and Catalyst switch-based routers.
.. The standalone gateways include the DT-24+, DE-30+, and VG200.
Router IOS-based gateway solutions are the 175x, 2600, 3600, 3810,
5300, 7200, and 7500.The switch-based gateways are the Catalyst 4000,
4200, and 6000 Series.These gateways run the following protocols:
H.323, MGCP, Skinny, and SIP.
www.syngress.com
98 Chapter 3 • AVVID Gateway Selection
.. The IP/VC 3500 family is the videoconferencing gateway products
from Cisco.
Introduction to AVVID Gateways
.. In the Cisco AVVID world, there are voice and video gateways to provide
connectivity to legacy networks. Cisco has voice gateways, which are
standalone routers, IOS-based routers, and Catalyst switch-based routers.
.. The standalone gateways include the DT-24+, DE-30+, and VG200.
Router IOS-based gateway solutions are the 175x, 2600, 3600, 3810,
5300, 7200, and 7500.The switch-based gateways are the Catalyst 4000,
4200, and 6000 Series.These gateways run the following protocols:
H.323, MGCP, Skinny, and SIP.
www.syngress.com
98 Chapter 3 • AVVID Gateway Selection
.. The IP/VC 3500 family is the videoconferencing gateway products
from Cisco.
Understanding the Capabilities of Gateway Protocols
Understanding the Capabilities of Gateway Protocols
.. H.323 is the most supported gateway protocol, backed by the Cisco
1750, 2600, 3600, AS5300, 7200, and 7500 Series routers.
.. Skinny Station Protocol allows a Skinny client to use TCP/IP to
transmit and receive calls as with DT-24+, DE-30+, and VG200.
.. MGCP is a master/slave protocol, where the gateway is the slave
servicing commands from the master, which is the call agent.The
MGCP protocol functions in an environment where the call control
intelligence have been removed from the gateway.
.. Session Initiation Protocol (SIP) is an application layer control protocol
that can establish, modify, and terminate multimedia sessions or calls.
.. H.323 is the most supported gateway protocol, backed by the Cisco
1750, 2600, 3600, AS5300, 7200, and 7500 Series routers.
.. Skinny Station Protocol allows a Skinny client to use TCP/IP to
transmit and receive calls as with DT-24+, DE-30+, and VG200.
.. MGCP is a master/slave protocol, where the gateway is the slave
servicing commands from the master, which is the call agent.The
MGCP protocol functions in an environment where the call control
intelligence have been removed from the gateway.
.. Session Initiation Protocol (SIP) is an application layer control protocol
that can establish, modify, and terminate multimedia sessions or calls.
AVVID Gateway Selection-The IP/VC 3500-Cisco 2600 route-Catalyst 4224.r
The importance of gateway selection is not to be overlooked, whether your
emphasis is on analog or digital protocols or both. Completely understanding all
the equipment’s features and benefits as well as the protocols should help make
this important decision easier to make.
AVVID voice gateways include standalone, IOS-based, and Catalyst switches.
The gateway protocols supported are H.323, MGCP, and Skinny with SIP
gaining ongoing popularity.The voice gateways range from small analog routers
such as the 1750 to large scalable digital T1/E1 7200 routers and everywhere in
between.The gateways can be more traditional VoIP toll bypass or total integrated
all-in-one solutions like the Catalyst 4224.
For small- to medium-sized organizations, the best solution may be either a
Cisco 2600 router or the Catalyst 4224. Either solution should not only be able
to handle VoIP solutions but other AVVID gateway requirements as well.The
2600 Series also has expansion capabilities to help with organizational growth.
However, if you do not require routing capabilities, you might look to the VG200
to provide similar solutions.When looking at the needs of a medium to large
organization, one would have to look at the 3600 Series router, which provides
the scalability necessary to handle the needs of a large enterprise environment.
The 3660 router has the ability to support up to 12 T1, which would consequently
support 2000+ users in a PSTN gateway scenario.The MC3810 would
provide a one step solution for data, voice and video needs. It provides VoFR,
VoIP but also VoATM. However, the MC3810 does not have the modular flexibility
of the 2600 or 3600 routers. It also does not integrate with CallManager.
When you are looking for switch-based solutions with similar functionality, as
the 2600/3600 Series routers do, the Catalyst 4000 would be a good choice. It
supports the same modules as the 2600/3600 except for the high-density voice
module, NM-HDV-XXX. For large organizations seeking high capacity and performance,
the choice could be the 7200 or 7500 routers with the ability to support
up to 20 T1s or 18 E1s via T1/E1 CAS or PRI signaling. A large-scale
switch-based solution would be the Catalyst 6500 series utilizing the 8-port
T1/E1 voice module. Since the release of the Catalyst 6513, which has 13 slots, it
could theoretically scale up to 96 T1 ports providing 2300+ voice channels. Most
likely the configuration would allocate some of the ports as T1 and others as DSP
resources, which will be discussed in Chapter 6.
The IP/VC 3500 videoconferencing products round out the gateways for
the AVVID architecture.They cover multipoint conference units, gateways, and
www.syngress.com
AVVID Gateway Selection • Chapter 3 97
terminal adapters.The IP/VC provides solutions in multipoint conferencing,
H.320 to H.323 translation, and legacy H.320 connectivity, while the MCM
completes the videoconferencing solution by providing the gatekeeper functionality
required for Video over IP.
The IP/VC 3510 MCU is a multiparticipant video and data conferencing
solution, whereas the IP/VC 3520 & 3525 gateways are used as ISDN H.320 to
IP H.323 gateways.The main difference between the latter two models is that the
3520 supports V.35 and ISDN BRI interfaces while ISDN PRI is available on the
3525.The IP/VC 3510 connects ISDN-based H.320 systems like a PictureTel
Venue 2000 to the IP-based H.323 network. Another MCU unit provided by
Cisco is the IP/VC 3540, which is a highly scalable multiparticipant videoconferencing
solution.The 3540 is targeted toward the large enterprise environments,
whereas, the 3510 is targeted to the low-end market.The IP/VC 3540 family of
products consists of a 3544 chassis, system module, MCU module, and application
server (AS) module, while the 3544 chassis has four slots with one required for a
system module and three designated for the other modules.The 3540 MCU is
available in three models: 30-, 60-, and 100-user types for system and nonsystems
modules.The MCM is part of Cisco IOS software, which runs on the Cisco
2500, 2600, 3600, 3810, and 7200 Series routers.The MCM function is to serve
as the videoconferencing gatekeeper and proxy.
Considering what’s been discussed in this chapter, you should now have a
greater understanding of the role gateway selection will have in developing your
ongoing enterprise solutions strategies, whether the importance lay in voice,
video, or both.
emphasis is on analog or digital protocols or both. Completely understanding all
the equipment’s features and benefits as well as the protocols should help make
this important decision easier to make.
AVVID voice gateways include standalone, IOS-based, and Catalyst switches.
The gateway protocols supported are H.323, MGCP, and Skinny with SIP
gaining ongoing popularity.The voice gateways range from small analog routers
such as the 1750 to large scalable digital T1/E1 7200 routers and everywhere in
between.The gateways can be more traditional VoIP toll bypass or total integrated
all-in-one solutions like the Catalyst 4224.
For small- to medium-sized organizations, the best solution may be either a
Cisco 2600 router or the Catalyst 4224. Either solution should not only be able
to handle VoIP solutions but other AVVID gateway requirements as well.The
2600 Series also has expansion capabilities to help with organizational growth.
However, if you do not require routing capabilities, you might look to the VG200
to provide similar solutions.When looking at the needs of a medium to large
organization, one would have to look at the 3600 Series router, which provides
the scalability necessary to handle the needs of a large enterprise environment.
The 3660 router has the ability to support up to 12 T1, which would consequently
support 2000+ users in a PSTN gateway scenario.The MC3810 would
provide a one step solution for data, voice and video needs. It provides VoFR,
VoIP but also VoATM. However, the MC3810 does not have the modular flexibility
of the 2600 or 3600 routers. It also does not integrate with CallManager.
When you are looking for switch-based solutions with similar functionality, as
the 2600/3600 Series routers do, the Catalyst 4000 would be a good choice. It
supports the same modules as the 2600/3600 except for the high-density voice
module, NM-HDV-XXX. For large organizations seeking high capacity and performance,
the choice could be the 7200 or 7500 routers with the ability to support
up to 20 T1s or 18 E1s via T1/E1 CAS or PRI signaling. A large-scale
switch-based solution would be the Catalyst 6500 series utilizing the 8-port
T1/E1 voice module. Since the release of the Catalyst 6513, which has 13 slots, it
could theoretically scale up to 96 T1 ports providing 2300+ voice channels. Most
likely the configuration would allocate some of the ports as T1 and others as DSP
resources, which will be discussed in Chapter 6.
The IP/VC 3500 videoconferencing products round out the gateways for
the AVVID architecture.They cover multipoint conference units, gateways, and
www.syngress.com
AVVID Gateway Selection • Chapter 3 97
terminal adapters.The IP/VC provides solutions in multipoint conferencing,
H.320 to H.323 translation, and legacy H.320 connectivity, while the MCM
completes the videoconferencing solution by providing the gatekeeper functionality
required for Video over IP.
The IP/VC 3510 MCU is a multiparticipant video and data conferencing
solution, whereas the IP/VC 3520 & 3525 gateways are used as ISDN H.320 to
IP H.323 gateways.The main difference between the latter two models is that the
3520 supports V.35 and ISDN BRI interfaces while ISDN PRI is available on the
3525.The IP/VC 3510 connects ISDN-based H.320 systems like a PictureTel
Venue 2000 to the IP-based H.323 network. Another MCU unit provided by
Cisco is the IP/VC 3540, which is a highly scalable multiparticipant videoconferencing
solution.The 3540 is targeted toward the large enterprise environments,
whereas, the 3510 is targeted to the low-end market.The IP/VC 3540 family of
products consists of a 3544 chassis, system module, MCU module, and application
server (AS) module, while the 3544 chassis has four slots with one required for a
system module and three designated for the other modules.The 3540 MCU is
available in three models: 30-, 60-, and 100-user types for system and nonsystems
modules.The MCM is part of Cisco IOS software, which runs on the Cisco
2500, 2600, 3600, 3810, and 7200 Series routers.The MCM function is to serve
as the videoconferencing gatekeeper and proxy.
Considering what’s been discussed in this chapter, you should now have a
greater understanding of the role gateway selection will have in developing your
ongoing enterprise solutions strategies, whether the importance lay in voice,
video, or both.
Gateway Selection Questions
Gateway Selection Questions
When designing and deploying a Cisco AVVID solution, the selection of
a gateway for the environment will most likely be based on several criteria.
The following is a sample list of questions regarding required features
that should be asked prior to selecting a gateway:
Is an analog or digital gateway required?
What is the required capacity of the gateway?
Is it a standalone or IOS-based gateway?
Do you want an integrated all-in-one solution?
Is IP routing required?
What type of connection is the gateway going to use (analog
POTS, PSTN, or PBX connection)?
What types of supplementary services are desired?
Is voice compression a part of the design? If so, which types?
Is direct inward dialing (DID) required?
Is calling line ID (CLID) needed?
Is fax relay needed?
What type of network management interface is preferred?
To which country will the hardware be shipped?
Is rack space available for all needed gateways, routers, and
switches?
The need to verify equipment features and capabilities is constant
as Cisco continues to update the IOS software features and routers/
gateways listed in this chapter
When designing and deploying a Cisco AVVID solution, the selection of
a gateway for the environment will most likely be based on several criteria.
The following is a sample list of questions regarding required features
that should be asked prior to selecting a gateway:
Is an analog or digital gateway required?
What is the required capacity of the gateway?
Is it a standalone or IOS-based gateway?
Do you want an integrated all-in-one solution?
Is IP routing required?
What type of connection is the gateway going to use (analog
POTS, PSTN, or PBX connection)?
What types of supplementary services are desired?
Is voice compression a part of the design? If so, which types?
Is direct inward dialing (DID) required?
Is calling line ID (CLID) needed?
Is fax relay needed?
What type of network management interface is preferred?
To which country will the hardware be shipped?
Is rack space available for all needed gateways, routers, and
switches?
The need to verify equipment features and capabilities is constant
as Cisco continues to update the IOS software features and routers/
gateways listed in this chapter
MCM Performance IP Routing
MCM Performance
IP Routing
Packet per H.323 Video Proxy
Platform Second Endpoints Video Calls Sessions
Cisco 7200 50–100K 3000 500 50 at 768 Kbps
75 at 384 Kbps
100 at 128 Kbps
Cisco 3660 25–100K 1800 250 25 at 768 Kbps
35 at 384 Kbps
50 at 128 Kbps
Cisco 3640 10–40K 1800 150 10 at 768 Kbps
15 at 384 Kbps
30 at 128 Kbps
Cisco 3620 10–15K 1800 75 10 at 768 Kbps
15 at 384 Kbps
30 at 128 Kbps
Cisco 262x 5–10K 900 60 2 at 768 Kbps
4 at 384 Kbps
6 at 128 Kbps
Cisco 261x 2–5K 900 60 2 at 768 Kbps
4 at 384 Kbps
6 at 128 Kbps
Cisco 3810 2–5K 900 60 2 at 768 Kbps
4 at 384 Kbps
6 at 128 Kbps
Cisco 25Xx N/A 600 30 2 at 768 Kbps
4 at 384 Kbps
10 at 128 Kbps
IP Routing
Packet per H.323 Video Proxy
Platform Second Endpoints Video Calls Sessions
Cisco 7200 50–100K 3000 500 50 at 768 Kbps
75 at 384 Kbps
100 at 128 Kbps
Cisco 3660 25–100K 1800 250 25 at 768 Kbps
35 at 384 Kbps
50 at 128 Kbps
Cisco 3640 10–40K 1800 150 10 at 768 Kbps
15 at 384 Kbps
30 at 128 Kbps
Cisco 3620 10–15K 1800 75 10 at 768 Kbps
15 at 384 Kbps
30 at 128 Kbps
Cisco 262x 5–10K 900 60 2 at 768 Kbps
4 at 384 Kbps
6 at 128 Kbps
Cisco 261x 2–5K 900 60 2 at 768 Kbps
4 at 384 Kbps
6 at 128 Kbps
Cisco 3810 2–5K 900 60 2 at 768 Kbps
4 at 384 Kbps
6 at 128 Kbps
Cisco 25Xx N/A 600 30 2 at 768 Kbps
4 at 384 Kbps
10 at 128 Kbps
30-session
30-session MCU module 30 participants at 128 Kbps, 15 participants
at 384 Kbps, 9 participants at 768 Kbps, 3 participants at 1.5/2.0 Mbps,
or 45 voice-only participants.
Based on an Intel Pentium server running Windows NT, the IP/VC 3540
Application Server (AS) acts as a T.120 data collaboration conferencing server
allowing end users to perform slide presentations, whiteboard, and other applications
during a conference call.The 3540 AS has the same participant user limits
as the MCU module.
The IP/VC 3540 H.32-to-H.323 Gateway module provides ISDN H.320 to
IP-based H.323 translation.This is the same functionality provided by the 3520 and
3525, and has two PRI ports that can be configured at T1 or E1 speeds.Along with
other IP/VC products, it supports H.261 and H.263 for video format encoding.
Voice encoding, meanwhile, is performed with G.711, G.722, and G.728.
at 384 Kbps, 9 participants at 768 Kbps, 3 participants at 1.5/2.0 Mbps,
or 45 voice-only participants.
Based on an Intel Pentium server running Windows NT, the IP/VC 3540
Application Server (AS) acts as a T.120 data collaboration conferencing server
allowing end users to perform slide presentations, whiteboard, and other applications
during a conference call.The 3540 AS has the same participant user limits
as the MCU module.
The IP/VC 3540 H.32-to-H.323 Gateway module provides ISDN H.320 to
IP-based H.323 translation.This is the same functionality provided by the 3520 and
3525, and has two PRI ports that can be configured at T1 or E1 speeds.Along with
other IP/VC products, it supports H.261 and H.263 for video format encoding.
Voice encoding, meanwhile, is performed with G.711, G.722, and G.728.
IP/VC 3510 MCU
IP/VC 3510 MCU
The IP/VC 3510 MCU merges three or more H.323 videoconference endpoints
into a single multiparticipant meeting (as shown in Figure 3.5).The 3510 MCU
is able to maintain ad hoc and scheduled videoconferences, and each unit can
support up to 15 sessions at 128 Kbps, nine sessions at 384 Kbps, seven sessions at
512 Kbps, five sessions at 768 Kbps, or three sessions at 1.5 Mbps.The unit can
support multiple conferences with a limiting factor of 15 sessions. Participants can
join through the Web interface, or by having the MCU dial to them.The MCU
can be cascaded together to support more sessions. In fact, the conference
capacity for multiple MCUs is 48 sessions.Videoconferencing, meanwhile, with
T.120 data sharing, is available along with audio only calls.
The IP/VC 3510 MCU merges three or more H.323 videoconference endpoints
into a single multiparticipant meeting (as shown in Figure 3.5).The 3510 MCU
is able to maintain ad hoc and scheduled videoconferences, and each unit can
support up to 15 sessions at 128 Kbps, nine sessions at 384 Kbps, seven sessions at
512 Kbps, five sessions at 768 Kbps, or three sessions at 1.5 Mbps.The unit can
support multiple conferences with a limiting factor of 15 sessions. Participants can
join through the Web interface, or by having the MCU dial to them.The MCU
can be cascaded together to support more sessions. In fact, the conference
capacity for multiple MCUs is 48 sessions.Videoconferencing, meanwhile, with
T.120 data sharing, is available along with audio only calls.
60-session
60-session MCU module 60 participants at 128 Kbps, 30 participants
at 384 Kbps, 15 participants at 768 Kbps, 5 participants at 1.5/2.0 Mbps,
or 150 voice-only participants.
at 384 Kbps, 15 participants at 768 Kbps, 5 participants at 1.5/2.0 Mbps,
or 150 voice-only participants.
100-session
100-session MCU module 100 participants at 128 Kbps, 50 participants
at 384 Kbps, 25 participants at 768 Kbps, 10 participants at 1.5/2.0
Mbps, or 150 voice-only participants.
at 384 Kbps, 25 participants at 768 Kbps, 10 participants at 1.5/2.0
Mbps, or 150 voice-only participants.
IP/VC 3540
IP/VC 3540
The IP/VC 3540 videoconferencing system is a multipoint conferencing,
gateway, and data collaboration integrated solution.The IP/VC 3540 solution
includes an IP/VC 3544 chassis and 3540 modules.The IP/VC 3544 chassis is a
2U 19 inch rack mountable unit that has four slots in a Compact PCI (cPCI) bus
backplane.The IP/VC 3540 system module manages the cPCI bus, and the
chassis supports two modules, an IP/VC 3540 MCU, Application Server, and
H.320-to-H.323 Gateway module.The management is performed through a Web
interface by java-enabled Web browsers.
The IP/VC 3540 MCU module supplies real-time voice, videoconferencing,
and T.120 data collaboration capabilities for companies desiring high quality and
scalability.The multipoint conferences can be scheduled or ad-hoc, while the
quality of video sessions range from 768 Kbps for high quality to 2 Mbps for
super-quality.The IP/VC 3540 MCU module supports multipoint videoconferences
with up to 100 participants, and comes in the following options: 30-, 60-,
and 100-sessions (128 Kbps).The following is a list of user limitations with performance
ratings:
The IP/VC 3540 videoconferencing system is a multipoint conferencing,
gateway, and data collaboration integrated solution.The IP/VC 3540 solution
includes an IP/VC 3544 chassis and 3540 modules.The IP/VC 3544 chassis is a
2U 19 inch rack mountable unit that has four slots in a Compact PCI (cPCI) bus
backplane.The IP/VC 3540 system module manages the cPCI bus, and the
chassis supports two modules, an IP/VC 3540 MCU, Application Server, and
H.320-to-H.323 Gateway module.The management is performed through a Web
interface by java-enabled Web browsers.
The IP/VC 3540 MCU module supplies real-time voice, videoconferencing,
and T.120 data collaboration capabilities for companies desiring high quality and
scalability.The multipoint conferences can be scheduled or ad-hoc, while the
quality of video sessions range from 768 Kbps for high quality to 2 Mbps for
super-quality.The IP/VC 3540 MCU module supports multipoint videoconferences
with up to 100 participants, and comes in the following options: 30-, 60-,
and 100-sessions (128 Kbps).The following is a list of user limitations with performance
ratings:
IP/VC 3520/3525 Gateway
IP/VC 3520/3525 Gateway
ISDN
IP WAN
IP/VC 3530
VTA
IP/VC 3525
Gateway
Switch
Figure 3.7 IP/VC 3530 VTA
IP/VC 3530
VTA
Session
IP Videoconferencing
Switch
ISDN
IP WAN
IP/VC 3530
VTA
IP/VC 3525
Gateway
Switch
Figure 3.7 IP/VC 3530 VTA
IP/VC 3530
VTA
Session
IP Videoconferencing
Switch
IP/VC 3530 VTA
IP/VC 3530 VTA
The IP/VC 3530 Video Terminal Adapter (VTA) is a 1U rack mounted unit with
one Ethernet port and two V.35 ports.The 3530 VTA (shown in Figure 3.7) connects
a single H.320 system to an IP network, and translates from a H.320 ISDNbased
network to a H.323 IP-based network.The video session can perform at
128 Kbps and up to 768 Kbps across the IP network, connecting to a multipoint
www.syngress.com
Figure 3.5 IP/VC 3510 MCU
IP Network
IP/VC 3510 MCU
AVVID Gateway Selection • Chapter 3 91
conference hosted by an IP/VC 3510 MCU by going through an IP/VC 3520 or
3525 gateway. Like all the other IP/VC products, it supports T.120 for data collaboration.
The encoding formats supported for video are H.261 and QCF/CIF,
while those supported for audio are G.711, G.722, and G.728.
The IP/VC 3530 Video Terminal Adapter (VTA) is a 1U rack mounted unit with
one Ethernet port and two V.35 ports.The 3530 VTA (shown in Figure 3.7) connects
a single H.320 system to an IP network, and translates from a H.320 ISDNbased
network to a H.323 IP-based network.The video session can perform at
128 Kbps and up to 768 Kbps across the IP network, connecting to a multipoint
www.syngress.com
Figure 3.5 IP/VC 3510 MCU
IP Network
IP/VC 3510 MCU
AVVID Gateway Selection • Chapter 3 91
conference hosted by an IP/VC 3510 MCU by going through an IP/VC 3520 or
3525 gateway. Like all the other IP/VC products, it supports T.120 for data collaboration.
The encoding formats supported for video are H.261 and QCF/CIF,
while those supported for audio are G.711, G.722, and G.728.
IP/VC 3510 MCU
IP/VC 3510 MCU
The Cisco IP/VC 3520 Gateway can bond to a maximum of three ports for
calls that require transfer rates of up to 384 Kbps. Each BRI interface allows a
128 Kbps call. Units with four BRI interface ports can simultaneously support
four 128 Kbps calls, two 256 Kbps calls, or one 384 Kbps call and one 128 Kbps
call. Each V.35 port supports transfer rates from 56 Kbps to 768 Kbps.The V.35
ports either RS-366 or V.25bis signaling. Both the 3520 and 3525 have an
embedded gatekeeper, where each supports video formats of H.261 and H.263.
The main differences between the two models are their interface connections
and the scalability of calls.
The 3525 gateway supports ISDN PRI T1 and E1 connections.The IP/VC
3525 T1 model supports up to three videoconferencing calls at 384 Kbps, while
the 3525 E1 model supports five calls at 384 Kbps. If less quality is acceptable, 13
calls at 128 Kbps is supported. Keep in mind, the 3525 supports the same audio
transcoding formats as the 3520 (Figure 3.6).
The Cisco IP/VC 3520 Gateway can bond to a maximum of three ports for
calls that require transfer rates of up to 384 Kbps. Each BRI interface allows a
128 Kbps call. Units with four BRI interface ports can simultaneously support
four 128 Kbps calls, two 256 Kbps calls, or one 384 Kbps call and one 128 Kbps
call. Each V.35 port supports transfer rates from 56 Kbps to 768 Kbps.The V.35
ports either RS-366 or V.25bis signaling. Both the 3520 and 3525 have an
embedded gatekeeper, where each supports video formats of H.261 and H.263.
The main differences between the two models are their interface connections
and the scalability of calls.
The 3525 gateway supports ISDN PRI T1 and E1 connections.The IP/VC
3525 T1 model supports up to three videoconferencing calls at 384 Kbps, while
the 3525 E1 model supports five calls at 384 Kbps. If less quality is acceptable, 13
calls at 128 Kbps is supported. Keep in mind, the 3525 supports the same audio
transcoding formats as the 3520 (Figure 3.6).
IP/VC 3520 and 3525 Gateway
IP/VC 3520 and 3525 Gateway
The IP/VC 3520 and 3525 provide translation services between H.320 and
H.323 networks.This allows companies to connect legacy ISDN H.320 videoconferencing
systems to IP-based H.323 networks, letting users conduct videoconferencing
across the IP LAN or via the PSTN.The IP/VC 3520 is available
in five different hardware configurations with the following features:
Four BRI interface ports
Four V.35 interface ports
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90 Chapter 3 • AVVID Gateway Selection
Combination of two BRI interface ports and two V.35 ports
Audio CODEC transcoding between G.711 and G.728
Audio CODEC transcoding between G.711 and G.723
Channel bonding
The IP/VC 3520 and 3525 provide translation services between H.320 and
H.323 networks.This allows companies to connect legacy ISDN H.320 videoconferencing
systems to IP-based H.323 networks, letting users conduct videoconferencing
across the IP LAN or via the PSTN.The IP/VC 3520 is available
in five different hardware configurations with the following features:
Four BRI interface ports
Four V.35 interface ports
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90 Chapter 3 • AVVID Gateway Selection
Combination of two BRI interface ports and two V.35 ports
Audio CODEC transcoding between G.711 and G.728
Audio CODEC transcoding between G.711 and G.723
Channel bonding
Choosing a Video Gateway Solution
Choosing a Video Gateway Solution
Cisco’s IP/VC 3500 family of products satisfies the video part of their AVVID
multiservice architecture.The IP/VC family covers video conferencing solutions
from the lower-end desktop to the high-end conferencing room implementation.
Considering today’s business environment and the never-ending effort to curb
expenses, companies are looking at more cost-effective ways of conducting business
meetings and presentations. One way is to institute a video conferencing
solution. In the following sections, we’ll discuss some of the products Cisco offers
in this area, including integrating with legacy video conferencing solutions.
Cisco’s IP/VC 3500 family of products satisfies the video part of their AVVID
multiservice architecture.The IP/VC family covers video conferencing solutions
from the lower-end desktop to the high-end conferencing room implementation.
Considering today’s business environment and the never-ending effort to curb
expenses, companies are looking at more cost-effective ways of conducting business
meetings and presentations. One way is to institute a video conferencing
solution. In the following sections, we’ll discuss some of the products Cisco offers
in this area, including integrating with legacy video conferencing solutions.
DPA 7630 Voice Mail Gateway PSTN Voice Mail Cisco
DPA 7630 Voice Mail Gateway
PSTN
Voice Mail
Cisco
CallManager
Voice Router
DPA 7630
IP Phone
IP Phone
Ethernet
Digital T1
(3) RJ21 Connections
Switch
AVVID Gateway Selection • Chapter 3 89
The DPA supports G.711 and G.729a voice CODECs to communicate
with the voice mail system, as well as the Avaya (formerly Lucent) Definity and
Meridian PBX along with the Octel voice mail system. Incoming/outgoing messages
and message waiting indicator (MWI) commands between CallManager and
the Octel voice mail are done through the DPA.
PSTN
Voice Mail
Cisco
CallManager
Voice Router
DPA 7630
IP Phone
IP Phone
Ethernet
Digital T1
(3) RJ21 Connections
Switch
AVVID Gateway Selection • Chapter 3 89
The DPA supports G.711 and G.729a voice CODECs to communicate
with the voice mail system, as well as the Avaya (formerly Lucent) Definity and
Meridian PBX along with the Octel voice mail system. Incoming/outgoing messages
and message waiting indicator (MWI) commands between CallManager and
the Octel voice mail are done through the DPA.
DPA 7610/7630 Voice Mail Gateway
DPA 7610/7630 Voice Mail Gateway
Another component of an AVVID solution is the DPA 7610/7630 Voice Mail
Gateway.The Digital PBX Adapter (DPA) 7610 and 7630 are VoIP gateways that
allow legacy voice mail systems to communicate with a Cisco CallManager.The
DPA 7610 and 7630 have a 10/100 Ethernet port and one or three RJ-21 ports,
respectively.The DPA uses the RJ-21 ports, which provide 8 to 24 four-wire digital
lines to interface with the legacy voice mail system. It communicates with the
CallManager through the Skinny Station protocol via the 10/100 Ethernet port,
and emulates an IP phone in order to communicate with the CallManager.The
DPA then allows simultaneous communication with a CallManager and legacy
PBX to the voice mail system (as illustrated in Figure 3.4).
Another component of an AVVID solution is the DPA 7610/7630 Voice Mail
Gateway.The Digital PBX Adapter (DPA) 7610 and 7630 are VoIP gateways that
allow legacy voice mail systems to communicate with a Cisco CallManager.The
DPA 7610 and 7630 have a 10/100 Ethernet port and one or three RJ-21 ports,
respectively.The DPA uses the RJ-21 ports, which provide 8 to 24 four-wire digital
lines to interface with the legacy voice mail system. It communicates with the
CallManager through the Skinny Station protocol via the 10/100 Ethernet port,
and emulates an IP phone in order to communicate with the CallManager.The
DPA then allows simultaneous communication with a CallManager and legacy
PBX to the voice mail system (as illustrated in Figure 3.4).
Cisco ICS-7750
Cisco ICS-7750
The System Processing Engine (SPE) 200 is the call-processing component,
which is a CallManager server on a blade.The module has an Intel Pentium II
CPU, 512MB of RAM, a 6.4GB hard drive, as well as Windows 2000 with SQL
Server and CallManager.
The Multiservice Route Processor (MRP) 200 is a voice-and-data-capable
router that can carry voice and data traffic over an IP network and can link
small-to-medium-size remote Ethernet LANs to central-offices LANs over different
types of WAN links.The MRP utilizes the same VIC,WIC, and VWIC
modules as the 1750, 2600, and 3600 series routers.The MRP 200 has a capacity
of two T1 ports for voice, as well as one port for data.
G.711 and G.729a are the voice CODECs used to support communication
between multiple IP and analog devices within campus and WAN environments.
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Figure 3.3 Cisco ICS-7750
WAN
PSTN
IP Phone
ICS-7750
IP Phone
Switch
88 Chapter 3 • AVVID Gateway Selection
These CODECs are achieved by the Packet Voice/fax DSP module (PVDM) in
the MRP.The MRP 200 has the capacity for 40-channel PVDM modules.
NOTE
Although the MRP 200 supports two T1 ports, it is limited to 24
simultaneous calls.
The System Processing Engine (SPE) 200 is the call-processing component,
which is a CallManager server on a blade.The module has an Intel Pentium II
CPU, 512MB of RAM, a 6.4GB hard drive, as well as Windows 2000 with SQL
Server and CallManager.
The Multiservice Route Processor (MRP) 200 is a voice-and-data-capable
router that can carry voice and data traffic over an IP network and can link
small-to-medium-size remote Ethernet LANs to central-offices LANs over different
types of WAN links.The MRP utilizes the same VIC,WIC, and VWIC
modules as the 1750, 2600, and 3600 series routers.The MRP 200 has a capacity
of two T1 ports for voice, as well as one port for data.
G.711 and G.729a are the voice CODECs used to support communication
between multiple IP and analog devices within campus and WAN environments.
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Figure 3.3 Cisco ICS-7750
WAN
PSTN
IP Phone
ICS-7750
IP Phone
Switch
88 Chapter 3 • AVVID Gateway Selection
These CODECs are achieved by the Packet Voice/fax DSP module (PVDM) in
the MRP.The MRP 200 has the capacity for 40-channel PVDM modules.
NOTE
Although the MRP 200 supports two T1 ports, it is limited to 24
simultaneous calls.
ICS 7750
ICS 7750
The Integrated Communication System (ICS) 7750 combines the features of a
multiservice router, a voice gateway, and call processing into a single chassis-based
solution.The ICS 7750 is not a typical VoIP gateway, but it’s covered because the
gateway module utilizes the same interface cards as some of the IOS-based gateways
discussed in this chapter.The ICS 7750 is targeted to branch offices and
midrange organizations—a typical installation supporting up to 150 users. A
complete IP telephony solution can be deployed with an ICS 7750, Catalyst
3524XL-PWR switch, and Cisco IP phones (as shown in Figure 3.3).
The Integrated Communication System (ICS) 7750 combines the features of a
multiservice router, a voice gateway, and call processing into a single chassis-based
solution.The ICS 7750 is not a typical VoIP gateway, but it’s covered because the
gateway module utilizes the same interface cards as some of the IOS-based gateways
discussed in this chapter.The ICS 7750 is targeted to branch offices and
midrange organizations—a typical installation supporting up to 150 users. A
complete IP telephony solution can be deployed with an ICS 7750, Catalyst
3524XL-PWR switch, and Cisco IP phones (as shown in Figure 3.3).
Catalyst 4224
Catalyst 4224
One of the newer switches to the Cisco line is the Catalyst 4200 Series family.
The Catalyst 4200 Access Gateway series switches allow voice, video, and data to
be offered to the small branch environment looking to deploy IP telephony solutions.
The Catalyst 4224 is a 2U rack mounted switch with 24 10/100 Ethernet
port and modular slots for voice and WAN modules.The same VIC and VWIC
modules mentioned previously for the Catalyst 4000 Series can be used in the
Catalyst 4224.
The 4200 Series was designed for small branch offices to deploy a complete
IP telephony solution, as well as enterprises in a centralized call processing
CallManager model.With a centralized call processing CallManager model if a
remote office loses its connection to the central office where the CallManager is
located, they will be unable to perform any voice calls from that office unless
they have a backup link to the central office or have Survivable Remote Site
(SRS) telephony software. If the company has several remote offices, it could be
rather expensive to have backup links for all these sites.
To provide a more cost-effective solution, Cisco developed SRS, which is part
of the IOS software that runs on a 4224 switch.As of this writing, SRS is also
available on Cisco 2600 and 3600 routers.This technology will also be included
in other Cisco products, such as the Cisco 175x router, and the Catalyst 4000
AGM by Q4 2001. SRS automatically detects a network failure, and using the
Cisco Simple Network Automated Provisioning (SNAP) capability, reconfigures
the router to provide call processing for the IP phones in that location.When the
WAN is restored, the router will shift call-processing functions back to the
CallManager cluster.
Smaller routers such as the 1750, 2600, and 3620 along with the Catalyst
4224 will support up to 24 phones. Meanwhile, the Cisco 3640 and Catalyst
4000 AGM support up to 48 phones, and the Cisco 3660 supports up to 144
phones. Up to two lines per phone are supported per system.
One of the newer switches to the Cisco line is the Catalyst 4200 Series family.
The Catalyst 4200 Access Gateway series switches allow voice, video, and data to
be offered to the small branch environment looking to deploy IP telephony solutions.
The Catalyst 4224 is a 2U rack mounted switch with 24 10/100 Ethernet
port and modular slots for voice and WAN modules.The same VIC and VWIC
modules mentioned previously for the Catalyst 4000 Series can be used in the
Catalyst 4224.
The 4200 Series was designed for small branch offices to deploy a complete
IP telephony solution, as well as enterprises in a centralized call processing
CallManager model.With a centralized call processing CallManager model if a
remote office loses its connection to the central office where the CallManager is
located, they will be unable to perform any voice calls from that office unless
they have a backup link to the central office or have Survivable Remote Site
(SRS) telephony software. If the company has several remote offices, it could be
rather expensive to have backup links for all these sites.
To provide a more cost-effective solution, Cisco developed SRS, which is part
of the IOS software that runs on a 4224 switch.As of this writing, SRS is also
available on Cisco 2600 and 3600 routers.This technology will also be included
in other Cisco products, such as the Cisco 175x router, and the Catalyst 4000
AGM by Q4 2001. SRS automatically detects a network failure, and using the
Cisco Simple Network Automated Provisioning (SNAP) capability, reconfigures
the router to provide call processing for the IP phones in that location.When the
WAN is restored, the router will shift call-processing functions back to the
CallManager cluster.
Smaller routers such as the 1750, 2600, and 3620 along with the Catalyst
4224 will support up to 24 phones. Meanwhile, the Cisco 3640 and Catalyst
4000 AGM support up to 48 phones, and the Cisco 3660 supports up to 144
phones. Up to two lines per phone are supported per system.
Catalyst 4000
Catalyst 4000
The Catalyst 4000 series is comprised of the 4003, 4006, 4908G, and 4912G
models.The Catalyst 4003 and 4006 are modular chassis-based switches, which are
a large part of the Cisco AVVID architecture.The Catalyst 4000 series family
(which is actually a scaled down version of the Catalyst 6000) is targeted at branch
offices, enterprise wiring closets, and mid-range organizations.The Catalyst 4006
supports inline-power Ethernet modules for Cisco IP phones while the Catalyst
4003 uses inline-power patch panels. It must use an external Auxiliary DC Power
Shelf to provide the needed power to the IP phones. For providing gateway services
to the IP telephony network, Cisco has an Access Gateway Module (AGM),
WS-X4604-GWY, for the Catalyst 4000.
The AGM allows the Catalyst 4000 to be an integrated solution providing IP
WAN routing, gateway functionality to the PSTN and PBX, and DSP resources
for CallManager. It is supported in both the 4003 and 4006, and uses the same
VIC and voice WAN interface card (VWIC) modules as the 1750, 2600, and 3600
series routers.The AGM has one dedicated VIC slot and two VWIC slots, which
holds either a VIC,VWIC, or WIC module.The AGM can be connected to the
PSTN or a PBX, and act as a H.323 gateway for CallManager.Analog devices
such as phones, speakerphones, and faxes can be connected to the AGM via an 8-
port RJ-21 FXS module,WS-U4604-8FXS, or one of the VIC card ports.The
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86 Chapter 3 • AVVID Gateway Selection
8-port FXS module is installed in the FlexSlot on the AGM, and the FlexSlot supplements
the other three VIC slots. In the future, the AGM will support a 16-port
FXS module allowing the AGM to support up to a total of 22 analog ports.The
three VIC slots, each holding a two-port card, provide the six other ports. FXS,
FXO, or E&M VIC modules can be used in these three VIC slots.
The Catalyst 4000 series is comprised of the 4003, 4006, 4908G, and 4912G
models.The Catalyst 4003 and 4006 are modular chassis-based switches, which are
a large part of the Cisco AVVID architecture.The Catalyst 4000 series family
(which is actually a scaled down version of the Catalyst 6000) is targeted at branch
offices, enterprise wiring closets, and mid-range organizations.The Catalyst 4006
supports inline-power Ethernet modules for Cisco IP phones while the Catalyst
4003 uses inline-power patch panels. It must use an external Auxiliary DC Power
Shelf to provide the needed power to the IP phones. For providing gateway services
to the IP telephony network, Cisco has an Access Gateway Module (AGM),
WS-X4604-GWY, for the Catalyst 4000.
The AGM allows the Catalyst 4000 to be an integrated solution providing IP
WAN routing, gateway functionality to the PSTN and PBX, and DSP resources
for CallManager. It is supported in both the 4003 and 4006, and uses the same
VIC and voice WAN interface card (VWIC) modules as the 1750, 2600, and 3600
series routers.The AGM has one dedicated VIC slot and two VWIC slots, which
holds either a VIC,VWIC, or WIC module.The AGM can be connected to the
PSTN or a PBX, and act as a H.323 gateway for CallManager.Analog devices
such as phones, speakerphones, and faxes can be connected to the AGM via an 8-
port RJ-21 FXS module,WS-U4604-8FXS, or one of the VIC card ports.The
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86 Chapter 3 • AVVID Gateway Selection
8-port FXS module is installed in the FlexSlot on the AGM, and the FlexSlot supplements
the other three VIC slots. In the future, the AGM will support a 16-port
FXS module allowing the AGM to support up to a total of 22 analog ports.The
three VIC slots, each holding a two-port card, provide the six other ports. FXS,
FXO, or E&M VIC modules can be used in these three VIC slots.
Catalyst 6000
Catalyst 6000
PSTN
Catalsyt 6000
with
Voice T1 and
FXS Modules
PBX
Cisco
CallManager
IP Phone
IP Phone
Analog
Phone
AVVID Gateway Selection • Chapter 3 85
It provides connectivity to legacy analog devices via a 24-port FXS module,
WS-X6624-FXS, which can be used for analog phones, faxes, speakerphones, and
modems.The analog FXS module acts as the gateway between the analog devices
and the IP network. By allowing for analog gateway functionality, organizations
can extend the useful life of their legacy analog devices. It also helps in the
migration to an IP telephony network by supporting existing fax machines and
conference speakerphones. Cisco does have an IP-based speakerphone, Cisco
7935, which was co-developed with Polycom.The analog to IP communication
is achieved via the voice CODECs of either G.711 or G.729a.
Digital PSTN and legacy PBX access is achieved with a Catalyst 6000 T1 or
E1 voice module, which is designed for larger enterprise campus environments.
The signaling supported for PSTN connections are Common channel signaling
(CCS) and ISDN PRI.The T1 module supports 23 channels in either signaling
mode, while the E1 module supports 29 channels for CCS and 30 for ISDN PRI
mode.The modules can also be configured to support transcoding and conference
bridging by configuring some ports for PBX or PSTN connectivity and
others for use as DSP resources.
PSTN
Catalsyt 6000
with
Voice T1 and
FXS Modules
PBX
Cisco
CallManager
IP Phone
IP Phone
Analog
Phone
AVVID Gateway Selection • Chapter 3 85
It provides connectivity to legacy analog devices via a 24-port FXS module,
WS-X6624-FXS, which can be used for analog phones, faxes, speakerphones, and
modems.The analog FXS module acts as the gateway between the analog devices
and the IP network. By allowing for analog gateway functionality, organizations
can extend the useful life of their legacy analog devices. It also helps in the
migration to an IP telephony network by supporting existing fax machines and
conference speakerphones. Cisco does have an IP-based speakerphone, Cisco
7935, which was co-developed with Polycom.The analog to IP communication
is achieved via the voice CODECs of either G.711 or G.729a.
Digital PSTN and legacy PBX access is achieved with a Catalyst 6000 T1 or
E1 voice module, which is designed for larger enterprise campus environments.
The signaling supported for PSTN connections are Common channel signaling
(CCS) and ISDN PRI.The T1 module supports 23 channels in either signaling
mode, while the E1 module supports 29 channels for CCS and 30 for ISDN PRI
mode.The modules can also be configured to support transcoding and conference
bridging by configuring some ports for PBX or PSTN connectivity and
others for use as DSP resources.
7200/7500 Signaling Protocols
7200/7500 Signaling Protocols
Signaling Protocol 7200 7500
CAS T1 12.0(5)XE3, 12.0(7)XK, 12.1(1)T* 12.1(3)T
CAS E1 12.0(5)XE3* 12.1(3)T
Q.SIG 12.0(7)XK*, 12.1(2)T 12.1(3)T
PRI Q.931 User Side 12.1(3)T 12.1(3)T
PRI Q.931 Network Side 12.1(3)T 12.1(3)T
R2 Signaling 12.1(3)T 12.1(3)T
Transparent CCS 12.1(3)T 12.1(3)T
Feature Group D 12.1(4)T 12.1(4)T
Multi-D Channel 12.1(3)T 12.1(3)T
RAI Future Future
NOTE
IOS Releases 12.1(3)T and earlier are supported on digital voice port
adapters PA-VXC-2TE1, PA-VXB-2TE1. IOS Releases 12.1(2)T and later are
supported on the enhanced digital voice port adapters PA-VXC-2TE1+
and PA-VXB-2TE1+.
The 7200 can perform as a high density H.323 AVVID gateway with a connection
to the PSTN and PBX, providing digital connection to both the PSTN
and PBX (as illustrated in Figure 3.1).
Signaling Protocol 7200 7500
CAS T1 12.0(5)XE3, 12.0(7)XK, 12.1(1)T* 12.1(3)T
CAS E1 12.0(5)XE3* 12.1(3)T
Q.SIG 12.0(7)XK*, 12.1(2)T 12.1(3)T
PRI Q.931 User Side 12.1(3)T 12.1(3)T
PRI Q.931 Network Side 12.1(3)T 12.1(3)T
R2 Signaling 12.1(3)T 12.1(3)T
Transparent CCS 12.1(3)T 12.1(3)T
Feature Group D 12.1(4)T 12.1(4)T
Multi-D Channel 12.1(3)T 12.1(3)T
RAI Future Future
NOTE
IOS Releases 12.1(3)T and earlier are supported on digital voice port
adapters PA-VXC-2TE1, PA-VXB-2TE1. IOS Releases 12.1(2)T and later are
supported on the enhanced digital voice port adapters PA-VXC-2TE1+
and PA-VXB-2TE1+.
The 7200 can perform as a high density H.323 AVVID gateway with a connection
to the PSTN and PBX, providing digital connection to both the PSTN
and PBX (as illustrated in Figure 3.1).
7200 T1/E1 Voice Port Adapters
7200 T1/E1 Voice Port Adapters
High-Complexity Medium-Complexity
Product Voice Channels Voice Channels
PA-VXB-2TE1+ 24 48
PA-VXC-2TE1+ 60 120
When installing multiple T1 or E1 voice port adapters, you must use a combination
of the PA-VXB-XXX and PA-MCX-XXX adapters in order to share
the DSP resources.Table 3.8 lists the Cisco IOS release supported on the given
platform for the given digital signaling protocol.
High-Complexity Medium-Complexity
Product Voice Channels Voice Channels
PA-VXB-2TE1+ 24 48
PA-VXC-2TE1+ 60 120
When installing multiple T1 or E1 voice port adapters, you must use a combination
of the PA-VXB-XXX and PA-MCX-XXX adapters in order to share
the DSP resources.Table 3.8 lists the Cisco IOS release supported on the given
platform for the given digital signaling protocol.
Cisco 7200/7500
Cisco 7200/7500
For enterprises seeking a high capacity and performance VoIP solution, the 7200
series routers are a viable choice.The Cisco 7200 with six slots can be equipped
with up to 20 T1s or 18 E1s, supporting voice-over-packet applications.The DS0
channel from a T1/E1 is switched into the Digital Signal Processor (DSP) to perform
the TDM-to-packet conversion of the bearer information present on a
DS0, while the Cisco 7200 router supports two types of port adapters, one with
DSPs and one without.The adapters without DSPs can use DSPs from the other
DSP-capable adapters.The PA-MCX-2/4/8 TE1 is a non-DSP adapter and
works with the PA-VXx-2TE1+ type adapters.The PA-VXx-2TE1+ adapters,
on the other hand, provide up to two T1 or E1 interfaces.Table 3.7 lists the
number of voice channels supported, based on the CODEC complexity.
For enterprises seeking a high capacity and performance VoIP solution, the 7200
series routers are a viable choice.The Cisco 7200 with six slots can be equipped
with up to 20 T1s or 18 E1s, supporting voice-over-packet applications.The DS0
channel from a T1/E1 is switched into the Digital Signal Processor (DSP) to perform
the TDM-to-packet conversion of the bearer information present on a
DS0, while the Cisco 7200 router supports two types of port adapters, one with
DSPs and one without.The adapters without DSPs can use DSPs from the other
DSP-capable adapters.The PA-MCX-2/4/8 TE1 is a non-DSP adapter and
works with the PA-VXx-2TE1+ type adapters.The PA-VXx-2TE1+ adapters,
on the other hand, provide up to two T1 or E1 interfaces.Table 3.7 lists the
number of voice channels supported, based on the CODEC complexity.
Cisco MC3810
Cisco MC3810
The Cisco MC3810 integrates data, voice, and video applications into a single
box solution.The MC3810 supports voice connectivity using the following
methods:Voice over Frame Relay (VoFR),Voice over ATM (VoATM), and VoIP.
The interfaces supported by the 3810 are the following:
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AVVID Gateway Selection • Chapter 3 81
Six analog ports (FXO, FXS, and E&M)
Digital T1/E1 (Drop and Insert, CAS, CCS, and PRI QSIG)
Ethernet 10Base-T
Two serial ports
Four BRI voice ports
The analog interfaces are not compatible or interchangeable with Cisco’s
1750, 2600, and 3600 routers.The MC3810 can interface with the PSTN or
PBX via digital connection.The digital T1 PBX connection supports 24 channels
for voice with the following compression CODECs: G.723.1, G.729, G.729a, and
G.726.This router provides similar Voice over X functionality as these routers, in
addition it has video capability such as circuit emulation over ATM, and H.323
gatekeeper for Video over IP.The MC3810 video features allow organizations to
get rid of H.320 ISDN dial-up circuits.
The Cisco MC3810 integrates data, voice, and video applications into a single
box solution.The MC3810 supports voice connectivity using the following
methods:Voice over Frame Relay (VoFR),Voice over ATM (VoATM), and VoIP.
The interfaces supported by the 3810 are the following:
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AVVID Gateway Selection • Chapter 3 81
Six analog ports (FXO, FXS, and E&M)
Digital T1/E1 (Drop and Insert, CAS, CCS, and PRI QSIG)
Ethernet 10Base-T
Two serial ports
Four BRI voice ports
The analog interfaces are not compatible or interchangeable with Cisco’s
1750, 2600, and 3600 routers.The MC3810 can interface with the PSTN or
PBX via digital connection.The digital T1 PBX connection supports 24 channels
for voice with the following compression CODECs: G.723.1, G.729, G.729a, and
G.726.This router provides similar Voice over X functionality as these routers, in
addition it has video capability such as circuit emulation over ATM, and H.323
gatekeeper for Video over IP.The MC3810 video features allow organizations to
get rid of H.320 ISDN dial-up circuits.
Catalyst 6000
Catalyst 6000
The Catalyst 6000 is an enterprise-class voice-capable switch, capable of supporting
analog and digital voice interfaces. It is a highly scalable switch, which makes it an
integral component of an AVVID network. As illustrated in Figure 3.2, the Catalyst
6000 can offer connectivity to the PSTN, legacy PBX, analog phones, and IP
phones. It provides inline power to Cisco IP phones via a 48-port 10/100 Ethernet
module,WS-X6348-RJ45V.
The Catalyst 6000 is an enterprise-class voice-capable switch, capable of supporting
analog and digital voice interfaces. It is a highly scalable switch, which makes it an
integral component of an AVVID network. As illustrated in Figure 3.2, the Catalyst
6000 can offer connectivity to the PSTN, legacy PBX, analog phones, and IP
phones. It provides inline power to Cisco IP phones via a 48-port 10/100 Ethernet
module,WS-X6348-RJ45V.
Cisco DT-24+/DE-30+
Cisco DT-24+/DE-30+
The DT-24/DE-30+ is a PCI-based digital gateway card that supports up to 23
or 30 voice channels.The card can be installed in a Cisco CallManager server or
any other PC with PCI slots, and only draws power from the PC.The DT-24/
DE-30+ card provides connectivity to the PSTN or a PBX.The DT-24/DE-30
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Figure 3.1 Cisco 7200
PSTN
PBX
Cisco
CallManager
IP Phone
IP Phone
Cisco 7200
H.323 Gateway
Switch
84 Chapter 3 • AVVID Gateway Selection
communicates with Cisco CallManager via Skinny Station Protocol and supports
supplementary services such as hold, transfer, and call forwarding.The DT-24/
DE-30 features ISDN PRI (T1/E1) rates for its trunk interface and a 10Base-T
Ethernet port for the IP network. If more trunks are needed, you can install as
many of the DT-24 or DE-30 cards that any given PC with PCI slots will support.
These cards provide similar functionality as the NM-HDV-1T1-24 or NMHDV-
1E1-30 modules for the Cisco 2600 and 3600 Series routers. If your company
has not invested in a router such as a 2600 or 3600 and has available PCs
with PCI slots, the DT-24 could be utilized to provide ISDN PRI connectivity. I
foresee the majority of development in further software and hardware enhancements
in the router and switch class gateways.This will allow enterprises to
leverage their current investment and knowledge in these platforms.
The DT-24/DE-30+ is a PCI-based digital gateway card that supports up to 23
or 30 voice channels.The card can be installed in a Cisco CallManager server or
any other PC with PCI slots, and only draws power from the PC.The DT-24/
DE-30+ card provides connectivity to the PSTN or a PBX.The DT-24/DE-30
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Figure 3.1 Cisco 7200
PSTN
PBX
Cisco
CallManager
IP Phone
IP Phone
Cisco 7200
H.323 Gateway
Switch
84 Chapter 3 • AVVID Gateway Selection
communicates with Cisco CallManager via Skinny Station Protocol and supports
supplementary services such as hold, transfer, and call forwarding.The DT-24/
DE-30 features ISDN PRI (T1/E1) rates for its trunk interface and a 10Base-T
Ethernet port for the IP network. If more trunks are needed, you can install as
many of the DT-24 or DE-30 cards that any given PC with PCI slots will support.
These cards provide similar functionality as the NM-HDV-1T1-24 or NMHDV-
1E1-30 modules for the Cisco 2600 and 3600 Series routers. If your company
has not invested in a router such as a 2600 or 3600 and has available PCs
with PCI slots, the DT-24 could be utilized to provide ISDN PRI connectivity. I
foresee the majority of development in further software and hardware enhancements
in the router and switch class gateways.This will allow enterprises to
leverage their current investment and knowledge in these platforms.
AS5800 Voice Feature Cards
AS5800 Voice Feature Cards
Part Number Description
DS58-336-MC-VOx AS5800 336-Port Medium Complexity Voice Card
DS58-192-MC-VOx AS5800 192-Port Medium Complexity Voice Card
DS58-192VOx AS5800 192-Port Voice Card
DS58-96VOx AS5800 96-Port Voice Card
Part Number Description
DS58-336-MC-VOx AS5800 336-Port Medium Complexity Voice Card
DS58-192-MC-VOx AS5800 192-Port Medium Complexity Voice Card
DS58-192VOx AS5800 192-Port Voice Card
DS58-96VOx AS5800 96-Port Voice Card
Cisco AS5300/AS5800
Cisco AS5300/AS5800
The AS5300 is a H.323-compliant enterprise-based VoIP gateway solution.The
AS5300 can scale to 96/120 connections based on the T1 or E1 modules installed.
This is accomplished with a quad T1 or E1 modules and two voice feature cards
that support 48/60 voice connections per card.
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The Cisco AS5800 supports up to 1344 voice ports in a single system, thus
offering a high concentration of VoIP DSPs in a single voice gateway.The Cisco
AS5800 supports two trunk cards: the 12-port T1/E1 and the channelized T3
termination card.The channelized T3 card provides 672 trunks, with a maximum
of two cards permitting 1344 trunks per AS5800 chassis. It supports voice feature
cards with a port density of 96 to 336 ports per card, as shown in Table 3.9.
The AS5300 is a H.323-compliant enterprise-based VoIP gateway solution.The
AS5300 can scale to 96/120 connections based on the T1 or E1 modules installed.
This is accomplished with a quad T1 or E1 modules and two voice feature cards
that support 48/60 voice connections per card.
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The Cisco AS5800 supports up to 1344 voice ports in a single system, thus
offering a high concentration of VoIP DSPs in a single voice gateway.The Cisco
AS5800 supports two trunk cards: the 12-port T1/E1 and the channelized T3
termination card.The channelized T3 card provides 672 trunks, with a maximum
of two cards permitting 1344 trunks per AS5800 chassis. It supports voice feature
cards with a port density of 96 to 336 ports per card, as shown in Table 3.9.
WARNING
WARNING
The commands listed for Under Dial Peers and Under Voice Ports should
not be configured in the MGCP gateway for MGCP-managed endpoints
(those with application MGCAPP command in their dial-peer statement).
It will cause communication problems with your CallManager.
8. The VG200 is configured to communicate with the CallManager server.
It will periodically send out messages attempting to establish a connection.
When the CallManager server configuration is complete, the connection
should automatically establish itself.
9. The following is the complete configuration for the VG200 router,
DNVRVG200A, for this document:
DNVRVG200A#show running-config
Building configuration...
Current configuration : 1244 bytes
!
version 12.1
no service single-slot-reload-enable
no service pad
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname DNVRVG200A
!
boot system flash
boot system rom
boot system tftp vg200 10.7.1.253
no logging buffered
logging rate-limit console 10 except errors
no logging console
enable secret jUie9834/#@skui
enable password #####
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!
ip subnet-zero
no ip finger
no ip domain-lookup
!
mgcp
mgcp call-agent 10.7.1.1
mgcp dtmf-relay codec all mode out-of-band
mgcp sdp simple
call rsvp-sync
!
!
!
ccm-manager mgcp
!
!
interface FastEthernet0/0
ip address 10.7.1.252 255.255.0.0
no ip mroute-cache
speed auto
full-duplex
!
ip default-gateway 10.7.1.254
ip classless
no ip http server
!
snmp-server engineID local 000000090200000196983000
snmp-server community public RO
!
voice-port 1/0/0
!
voice-port 1/0/1
!
voice-port 1/1/0
!
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80 Chapter 3 • AVVID Gateway Selection
voice-port 1/1/1
!
dial-peer voice 1 pots
application mgcpapp
port 1/0/0
!
dial-peer voice 2 pots
application mgcpapp
port 1/0/1
!
dial-peer voice 3 pots
application mgcpapp
port 1/1/0
!
dial-peer voice 4 pots
application mgcpapp
port 1/1/1
!
!
line con 0
transport input none
line aux 0
line vty 0 4
password ww
login
!
end
DNVRVG200A#
The commands listed for Under Dial Peers and Under Voice Ports should
not be configured in the MGCP gateway for MGCP-managed endpoints
(those with application MGCAPP command in their dial-peer statement).
It will cause communication problems with your CallManager.
8. The VG200 is configured to communicate with the CallManager server.
It will periodically send out messages attempting to establish a connection.
When the CallManager server configuration is complete, the connection
should automatically establish itself.
9. The following is the complete configuration for the VG200 router,
DNVRVG200A, for this document:
DNVRVG200A#show running-config
Building configuration...
Current configuration : 1244 bytes
!
version 12.1
no service single-slot-reload-enable
no service pad
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname DNVRVG200A
!
boot system flash
boot system rom
boot system tftp vg200 10.7.1.253
no logging buffered
logging rate-limit console 10 except errors
no logging console
enable secret jUie9834/#@skui
enable password #####
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!
ip subnet-zero
no ip finger
no ip domain-lookup
!
mgcp
mgcp call-agent 10.7.1.1
mgcp dtmf-relay codec all mode out-of-band
mgcp sdp simple
call rsvp-sync
!
!
!
ccm-manager mgcp
!
!
interface FastEthernet0/0
ip address 10.7.1.252 255.255.0.0
no ip mroute-cache
speed auto
full-duplex
!
ip default-gateway 10.7.1.254
ip classless
no ip http server
!
snmp-server engineID local 000000090200000196983000
snmp-server community public RO
!
voice-port 1/0/0
!
voice-port 1/0/1
!
voice-port 1/1/0
!
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80 Chapter 3 • AVVID Gateway Selection
voice-port 1/1/1
!
dial-peer voice 1 pots
application mgcpapp
port 1/0/0
!
dial-peer voice 2 pots
application mgcpapp
port 1/0/1
!
dial-peer voice 3 pots
application mgcpapp
port 1/1/0
!
dial-peer voice 4 pots
application mgcpapp
port 1/1/1
!
!
line con 0
transport input none
line aux 0
line vty 0 4
password ww
login
!
end
DNVRVG200A#
VG-200
VG-200
The VG-200 is a VoIP gateway which is modular, 1U rack-mounted, and based
on the Cisco 2600 chassis.The VG-200 is similar to the Cisco 2600, but without
routing capabilities. It supports connectivity to PBX, PSTN, analog and digital
dial access, and legacy voice mail systems, and has the capacity for up to four
analog ports or two digital ports. A common configuration of a small branch
office is the VG-200 with a NM-2V module and two VIC cards.The VIC cards
can be FXS, FXO, E&M, or ISDN Basic Rate Interface (BRI). An upgraded configuration
would have one digital T1/E1 voice module (NM-HDV), which has
either 1 or 2 ports.This configuration supports up to 60 voice channels.The VG-
200 communicates with Cisco CallManager via MGCP and H.323 protocol.
FXS and FXO are the only ports supported under the MGCP configuration
mode with CallManager.The next section contains an example configuration of
how you might implement MGCP in a small office environment using a VG200.
This example illustrates the use of multiple FXO and FXS ports.The FXS ports
could be used for a legacy conference speakerphone and fax machine with the
FXO ports providing PSTN connectivity.
The VG-200 is a VoIP gateway which is modular, 1U rack-mounted, and based
on the Cisco 2600 chassis.The VG-200 is similar to the Cisco 2600, but without
routing capabilities. It supports connectivity to PBX, PSTN, analog and digital
dial access, and legacy voice mail systems, and has the capacity for up to four
analog ports or two digital ports. A common configuration of a small branch
office is the VG-200 with a NM-2V module and two VIC cards.The VIC cards
can be FXS, FXO, E&M, or ISDN Basic Rate Interface (BRI). An upgraded configuration
would have one digital T1/E1 voice module (NM-HDV), which has
either 1 or 2 ports.This configuration supports up to 60 voice channels.The VG-
200 communicates with Cisco CallManager via MGCP and H.323 protocol.
FXS and FXO are the only ports supported under the MGCP configuration
mode with CallManager.The next section contains an example configuration of
how you might implement MGCP in a small office environment using a VG200.
This example illustrates the use of multiple FXO and FXS ports.The FXS ports
could be used for a legacy conference speakerphone and fax machine with the
FXO ports providing PSTN connectivity.
Cisco 3600
Cisco 3600
The 3600 Series router covers the following models 362x, 364x, and 366x.The
3600 Series routers are similar to the 2600 Series.The main differences between
the 2600 and 3600 are scalability and performance.Table 3.6 lists the different
models and their capacity of analog and digital channels.The Cisco 3660 Voice
Gateway interoperates with H.323-compliant voice and videoconferencing applications
such as Microsoft NetMeeting, as well as third-party H.323-compliant
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gateways and gatekeepers.The 3660 router is a high-performance (100 mips)
gateway and gatekeeper, which can be used in an AVVID CallManager deployment
or in a toll bypass VoIP environment.
The 3600 Series router covers the following models 362x, 364x, and 366x.The
3600 Series routers are similar to the 2600 Series.The main differences between
the 2600 and 3600 are scalability and performance.Table 3.6 lists the different
models and their capacity of analog and digital channels.The Cisco 3660 Voice
Gateway interoperates with H.323-compliant voice and videoconferencing applications
such as Microsoft NetMeeting, as well as third-party H.323-compliant
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AVVID Gateway Selection • Chapter 3 75
gateways and gatekeepers.The 3660 router is a high-performance (100 mips)
gateway and gatekeeper, which can be used in an AVVID CallManager deployment
or in a toll bypass VoIP environment.
Cisco 3600 Series Analog/Digital Scalability
Cisco 3600 Series Analog/Digital Scalability
Digital PSTN
Model Slots Analog PSTN DS0s T1/E1 Performance (pps)
Cisco 3620 2 4 48/60 20–40,000
Cisco 3640 4 12 136/180 50–70,000
Cisco 3660 6 24 288/360 120,000
Digital PSTN
Model Slots Analog PSTN DS0s T1/E1 Performance (pps)
Cisco 3620 2 4 48/60 20–40,000
Cisco 3640 4 12 136/180 50–70,000
Cisco 3660 6 24 288/360 120,000
Cisco 2600 Series Routers
Cisco 2600 Series Routers
Model Interface Processor Performance (pps)
2610 One 10 Mbps Ethernet 40MHz 15,000
2611 Two 10 Mbps Ethernet 40MHz 15,000
2612 One 10 Mbps Ethernet 40MHz
One Token Ring 15,000
2613 Two Token Ring 40MHz 15,000
2620 One 10/100 Mbps Ethernet 50MHz 25,000
2621 Two 10/100 Mbps Ethernet 50MHz 25,000
2650 One 10/100 Mbps Ethernet 80MHz 37,000
2651 Two 10/100 Mbps Ethernet 80MHz 37,000
All the 2600 Series routers provide the same number of network module slots
and WAN interface slots, which is one and two, respectively.The network module
slot can hold one two-slot voice/fax network module or high-density voice/fax
module.The two-slot voice/fax network module allows for two of the two-port
FXS, FXO, and E&M voice interface cards or any combination thereof.The
high-density voice modules, meanwhile, support one to two T1 or E1 ports.
The Cisco 2600 can be used in a traditional VoIP environment providing tollbypass,
or be integrated into an AVVID architecture.The 2600 router communicates
with CallManager via H.323 and MGCP protocol. As of CallManager
3.0(5), MGCP protocol is supported on FXS and FXO analog interfaces. A small
company with a 25-user site could use a 2600 router with two two-port FXO
modules providing four analog lines for outgoing and incoming calls.This router
would also provide IP WAN connectivity. As the site grows, the Cisco 2600 can
be upgraded from the two two-port FXO modules to a high-density voice
module, NM-HDV-1T1-24, which will provide more trunks for the user base in
this larger environment.
Model Interface Processor Performance (pps)
2610 One 10 Mbps Ethernet 40MHz 15,000
2611 Two 10 Mbps Ethernet 40MHz 15,000
2612 One 10 Mbps Ethernet 40MHz
One Token Ring 15,000
2613 Two Token Ring 40MHz 15,000
2620 One 10/100 Mbps Ethernet 50MHz 25,000
2621 Two 10/100 Mbps Ethernet 50MHz 25,000
2650 One 10/100 Mbps Ethernet 80MHz 37,000
2651 Two 10/100 Mbps Ethernet 80MHz 37,000
All the 2600 Series routers provide the same number of network module slots
and WAN interface slots, which is one and two, respectively.The network module
slot can hold one two-slot voice/fax network module or high-density voice/fax
module.The two-slot voice/fax network module allows for two of the two-port
FXS, FXO, and E&M voice interface cards or any combination thereof.The
high-density voice modules, meanwhile, support one to two T1 or E1 ports.
The Cisco 2600 can be used in a traditional VoIP environment providing tollbypass,
or be integrated into an AVVID architecture.The 2600 router communicates
with CallManager via H.323 and MGCP protocol. As of CallManager
3.0(5), MGCP protocol is supported on FXS and FXO analog interfaces. A small
company with a 25-user site could use a 2600 router with two two-port FXO
modules providing four analog lines for outgoing and incoming calls.This router
would also provide IP WAN connectivity. As the site grows, the Cisco 2600 can
be upgraded from the two two-port FXO modules to a high-density voice
module, NM-HDV-1T1-24, which will provide more trunks for the user base in
this larger environment.
Cisco 2600
Cisco 2600
The 2600 and 3600 Series routers are the mid-range multiservice platform
routers.The 2600 series is comprised of ten models, three performance levels, and
three topology types (listed in Table 3.5).
www
The 2600 and 3600 Series routers are the mid-range multiservice platform
routers.The 2600 series is comprised of ten models, three performance levels, and
three topology types (listed in Table 3.5).
www
Cisco 1750
Cisco 1750
The Cisco 1750 is an entry-level multiservice router supporting VoIP.The 1750
router can support 2 to 4 analog voice ports, as well as the FXS, FXO, and E&M
voice interface cards. It is typically used as a small-scale toll bypass solution.These
voice interface cards are the same modules used with the Cisco 2600 and 3600
series routers.VoIP call admission via RSVP has been supported on the 1750
since release 12.1(3)XI.
Cisco IOS Release 12.1(3)T supports the following features:
Low Latency Queuing for Voice over Frame Relay
H.323 voice (v2) support
RAS protocol voice (v2) enhancement support
DiffServ
Fax relay enhancements
Another new feature supported in 12.1(5)XM or later is digital CAS (E&M)
interfaces in addition to the analog (FXO, FXS, and E&M) interfaces.The 1750
router is not a typical gateway in a CallManager environment. Again, the Cisco
1750 is a good fit for the small office environment deploying a toll-bypass solution.
For example, let’s imagine we have a company with a main office in a
metropolitan area and have several small field offices. In the field offices, the 1750
could be used to connect to a key system with a FXO port on the switch trunk
side and have WAN connectivity back to the main office.This would allow the
several field offices to utilize their current WAN circuits for voice and data,
which would eliminate long distance charges back to headquarters.
The Cisco 1750 is an entry-level multiservice router supporting VoIP.The 1750
router can support 2 to 4 analog voice ports, as well as the FXS, FXO, and E&M
voice interface cards. It is typically used as a small-scale toll bypass solution.These
voice interface cards are the same modules used with the Cisco 2600 and 3600
series routers.VoIP call admission via RSVP has been supported on the 1750
since release 12.1(3)XI.
Cisco IOS Release 12.1(3)T supports the following features:
Low Latency Queuing for Voice over Frame Relay
H.323 voice (v2) support
RAS protocol voice (v2) enhancement support
DiffServ
Fax relay enhancements
Another new feature supported in 12.1(5)XM or later is digital CAS (E&M)
interfaces in addition to the analog (FXO, FXS, and E&M) interfaces.The 1750
router is not a typical gateway in a CallManager environment. Again, the Cisco
1750 is a good fit for the small office environment deploying a toll-bypass solution.
For example, let’s imagine we have a company with a main office in a
metropolitan area and have several small field offices. In the field offices, the 1750
could be used to connect to a key system with a FXO port on the switch trunk
side and have WAN connectivity back to the main office.This would allow the
several field offices to utilize their current WAN circuits for voice and data,
which would eliminate long distance charges back to headquarters.
Configuring and Installing a VG200 with MGCP
Configuring and Installing a VG200 with MGCP
The following is the step-by-step procedure to configure a VG200 with MGCP
as its gateway protocol.
1. First, configure an IP address on the VG200’s Ethernet interface and
enable the interface:
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76 Chapter 3 • AVVID Gateway Selection
router(config)#interface fastethernet 0/0
router(config-if)#ip address 10.7.1.252 255.255.255.0
router(config-if)#no shut
2. Next, assign a unique name to the VG200 so the CallManager server can
identify it:
router(config)#hostname DNVRVG200a
This is how CallManager keeps track of the MGCP network devices
it is communicating with.This name must be unique.
3. Configure the VG200 to run MGCP as a signaling protocol.
DNVRVG200A(config)#mgcp
4. Configure the IP address (or DNS Name) for the CallManager server:
VG200(config)#mgcp call-agent 10.7.1.1
5. Select the CODEC type and the DTMF relay function:
DNVRVG200A(config)#mgcp dtmf-relay codec all mode out-of-band
6. To enable support for Cisco CallManager within MGCP, enter the
following command:
DNVRVG200A(config)#ccm-manager mgcp
NOTE
Use the command show voice port to determine the type of ports the
VG200 has and which order they are installed in.
7. Bind the MGCP application to the voice ports.
FXO Port:
DNVRVG200A(config)#dial-peer voice 1 pots
DNVRVG200A(config)#application MGCPAPP
DNVRVG200A(config)#port 1/0/0
FXO Port:
DNVRVG200A(config)#dial-peer voice 2 pots
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DNVRVG200A(config)#application MGCPAPP
DNVRVG200A(config)#port 1/0/1
FXS Port:
DNVRVG200A(config)#dial-peer voice 3 pots
DNVRVG200A(config)#application MGCPAPP
DNVRVG200A(config)#port 1/1/0
FXS Port:
DNVRVG200A(config)#dial-peer voice 4 pots
DNVRVG200A(config)#application MGCPAPP
DNVRVG200A(config)#port 1/1/1
NOTE
In some Cisco IOS versions, the application MGCPAPP command is casesensitive.
As a precaution, always enter this command in uppercase.
Make certain the voice ports are enabled as well. Executing the command
no shut enables both ports on a VIC.
FXO Port:
DNVRVG200A(config)#voice-port 1/0/0
DNVRVG200A(config-voiceport)#no shut
FXS Port:
DNVRVG200A(config)#voice-port 1/1/0
DNVRVG200A(config-voiceport)#no shut
Under Dial Peers:
destination-pattern
session-target
Under Voice Ports:
connection { plar | tie line | trunk }
The following is the step-by-step procedure to configure a VG200 with MGCP
as its gateway protocol.
1. First, configure an IP address on the VG200’s Ethernet interface and
enable the interface:
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76 Chapter 3 • AVVID Gateway Selection
router(config)#interface fastethernet 0/0
router(config-if)#ip address 10.7.1.252 255.255.255.0
router(config-if)#no shut
2. Next, assign a unique name to the VG200 so the CallManager server can
identify it:
router(config)#hostname DNVRVG200a
This is how CallManager keeps track of the MGCP network devices
it is communicating with.This name must be unique.
3. Configure the VG200 to run MGCP as a signaling protocol.
DNVRVG200A(config)#mgcp
4. Configure the IP address (or DNS Name) for the CallManager server:
VG200(config)#mgcp call-agent 10.7.1.1
5. Select the CODEC type and the DTMF relay function:
DNVRVG200A(config)#mgcp dtmf-relay codec all mode out-of-band
6. To enable support for Cisco CallManager within MGCP, enter the
following command:
DNVRVG200A(config)#ccm-manager mgcp
NOTE
Use the command show voice port to determine the type of ports the
VG200 has and which order they are installed in.
7. Bind the MGCP application to the voice ports.
FXO Port:
DNVRVG200A(config)#dial-peer voice 1 pots
DNVRVG200A(config)#application MGCPAPP
DNVRVG200A(config)#port 1/0/0
FXO Port:
DNVRVG200A(config)#dial-peer voice 2 pots
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DNVRVG200A(config)#application MGCPAPP
DNVRVG200A(config)#port 1/0/1
FXS Port:
DNVRVG200A(config)#dial-peer voice 3 pots
DNVRVG200A(config)#application MGCPAPP
DNVRVG200A(config)#port 1/1/0
FXS Port:
DNVRVG200A(config)#dial-peer voice 4 pots
DNVRVG200A(config)#application MGCPAPP
DNVRVG200A(config)#port 1/1/1
NOTE
In some Cisco IOS versions, the application MGCPAPP command is casesensitive.
As a precaution, always enter this command in uppercase.
Make certain the voice ports are enabled as well. Executing the command
no shut enables both ports on a VIC.
FXO Port:
DNVRVG200A(config)#voice-port 1/0/0
DNVRVG200A(config-voiceport)#no shut
FXS Port:
DNVRVG200A(config)#voice-port 1/1/0
DNVRVG200A(config-voiceport)#no shut
Under Dial Peers:
destination-pattern
session-target
Under Voice Ports:
connection { plar | tie line | trunk }
Introduction to AVVID Gateways
Introduction to AVVID Gateways
A gateway, by definition, is a device that converts one media or protocol to
another. In the AVVID or Voice over IP (VoIP) environment, a gateway is responsible
for connecting an IP telephone network to the PSTN or PBX and key systems.
For example, the gateway may connect an H.323 network to an SIP-based
network, PSTN, or ISDN. It also performs translations between different transmission
formats and communication procedures, and is responsible for setting up
and clearing calls on both sides. Communication between terminals and gateways
is done through the H.245 and Q.931 protocols.
Types of gateways range from specialized entry-level standalone devices to
enterprise-level integrated router and switch gateways. Based on the device or
the implementation, the gateways communicate with Cisco CallManager or
other network devices over various gateway protocols.Your own infrastructure
and VoIP requirements will help determine what gateway is right for you, but
required common features include: DTMF relay, CallManager redundancy, and
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supplementary services. Supplementary services allow users to perform call hold,
transfer, and conferencing.
AVVID gateways themselves are in the form of analog and digital versions
running different protocols, each of which we’ll cover in the coming sections.
A gateway, by definition, is a device that converts one media or protocol to
another. In the AVVID or Voice over IP (VoIP) environment, a gateway is responsible
for connecting an IP telephone network to the PSTN or PBX and key systems.
For example, the gateway may connect an H.323 network to an SIP-based
network, PSTN, or ISDN. It also performs translations between different transmission
formats and communication procedures, and is responsible for setting up
and clearing calls on both sides. Communication between terminals and gateways
is done through the H.245 and Q.931 protocols.
Types of gateways range from specialized entry-level standalone devices to
enterprise-level integrated router and switch gateways. Based on the device or
the implementation, the gateways communicate with Cisco CallManager or
other network devices over various gateway protocols.Your own infrastructure
and VoIP requirements will help determine what gateway is right for you, but
required common features include: DTMF relay, CallManager redundancy, and
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supplementary services. Supplementary services allow users to perform call hold,
transfer, and conferencing.
AVVID gateways themselves are in the form of analog and digital versions
running different protocols, each of which we’ll cover in the coming sections.
Understanding the Capabilities of Gateway Protocols
Understanding the Capabilities
of Gateway Protocols
The three voice gateway protocols supported in Cisco’s AVVID architecture are
Skinny Station Protocol (SSP),H.323, and MGCP. Skinny Station Protocol allows a
Skinny client to use TCP/IP to transmit and receive calls and RTP/UDP/IP packets
for audio.An example of a Skinny client is an IP phone or gateway. The Skinny
clients communicate with a Cisco CallManager over TCP on ports 2000–2002. SSP
was developed by Cisco (formerly Selsius) as a low-bandwidth gateway protocol.
H.323 is the most supported gateway protocol from Cisco, and is an ITU-T
(Telecommunication Standardization Sector of the International Telecommunications
Union) standard for packet-based audio, video, and conferencing. It is the
umbrella standard for the conferencing standard (made up of others such as H.245,
H.225, and Q.931), and is the only gateway that provides full routing capabilities. It
transmits and receives media streams via RTP with Real-Time Control Protocol
(RTCP), carried over UDP, thereby providing status and control information.
Signaling, such as Registration,Admission, and Status (RAS), H.245, and Q.931 is
transported over TCP.Q.931 signaling, meanwhile, is for call setup and termination.
Capabilities, however, are exchanged by utilizing H.245, which is for call control,
and establishes multimedia communication or call services between the H.323
clients. Some new features to be supported in H.323v2 are H.235, H.450.x, Fast
Connect, alternative gatekeepers,Q.931 forwarding, and integration of T.120. For
its part, H.235 allows security and authentication, such as registration passwords,
while H.450.x offers supplementary services.
The MGCP protocol functions in an architecture where the call control
intelligence is removed from the gateway. Level3, Bellcore, Cisco, and Nortel
developed MGCP (described in Request for Comments [RFC] 2705), which is a
master/slave protocol, where the gateway is the slave servicing commands from
the master, which is the call agent. In the Cisco AVVID environment, the
CallManager functions as the call agent.Two benefits MGCP provides over
H.323 are centralized dial plans and dynamic dial plan updates versus statically
configuring each H.323 gateway. (The MGCP protocol communicates over UDP
port 2427 and TCP port 2428.)
AVVID Gateway Selection • Chapter 3 67
68 Chapter 3 • AVVID Gateway Selection
The skinny gateways are the DT-24+, DE-30+, and Catalyst 4000/6000
modules, which provide CallManager access to digital gateways. An example of
an H.323 gateway is a Cisco IOS router like the 2600 and 3600.The VG-200 is
an MGCP gateway with future support for the 2600, 3600, 3810, and Catalyst
modules.
Another protocol being implemented in Cisco gateways is the Session
Initiation Protocol. SIP (described in RFC 2543) is an application-layer control
protocol that can establish, modify, and terminate multimedia sessions or calls.
These sessions include IP conferences, telephone calls, and multimedia distribution.
A Cisco VoIP solution for SIP consists of a SIP agent, 7960 IP Phone, SIP
gateway, and a SIP proxy server.
SIP supports five elements of establishing and terminating communications:
User location
User capabilities
User availability
Call setup
Call handling
Currently, the VoIP world is dominated by H.323; the emergence of SIP and
the increasing number of applications supporting this new technology means the
interoperability of SIP with existing H.323 networks. SIP was first available in
IOS version 12.1(1)T. SIP was enhanced in IOS version 12.2(2)XA, supporting
the Cisco 2600 and 3600 Series router-voice gateways, and the Cisco AS5300,
AS5350, and AS5400 access server-voice gateways now allow both SIP and H.323
to fully coexist, as well as allowing interoperability between the two protocols.
NOTE
An example of new software utilizing the functionality of SIP is the application
Windows Messenger, which is part of Windows XP. Windows
Messenger is real-time communications software that provides end-toend
IP telephony. SIP is a multipart communication protocol, but the first
version of Windows Messenger will only support a two-way conversation.
Video gateways are used to convert form H.320 devices to H.323 devices.
The Cisco IP/VC products allow companies to utilize their legacy H.320 ISDNwww
of Gateway Protocols
The three voice gateway protocols supported in Cisco’s AVVID architecture are
Skinny Station Protocol (SSP),H.323, and MGCP. Skinny Station Protocol allows a
Skinny client to use TCP/IP to transmit and receive calls and RTP/UDP/IP packets
for audio.An example of a Skinny client is an IP phone or gateway. The Skinny
clients communicate with a Cisco CallManager over TCP on ports 2000–2002. SSP
was developed by Cisco (formerly Selsius) as a low-bandwidth gateway protocol.
H.323 is the most supported gateway protocol from Cisco, and is an ITU-T
(Telecommunication Standardization Sector of the International Telecommunications
Union) standard for packet-based audio, video, and conferencing. It is the
umbrella standard for the conferencing standard (made up of others such as H.245,
H.225, and Q.931), and is the only gateway that provides full routing capabilities. It
transmits and receives media streams via RTP with Real-Time Control Protocol
(RTCP), carried over UDP, thereby providing status and control information.
Signaling, such as Registration,Admission, and Status (RAS), H.245, and Q.931 is
transported over TCP.Q.931 signaling, meanwhile, is for call setup and termination.
Capabilities, however, are exchanged by utilizing H.245, which is for call control,
and establishes multimedia communication or call services between the H.323
clients. Some new features to be supported in H.323v2 are H.235, H.450.x, Fast
Connect, alternative gatekeepers,Q.931 forwarding, and integration of T.120. For
its part, H.235 allows security and authentication, such as registration passwords,
while H.450.x offers supplementary services.
The MGCP protocol functions in an architecture where the call control
intelligence is removed from the gateway. Level3, Bellcore, Cisco, and Nortel
developed MGCP (described in Request for Comments [RFC] 2705), which is a
master/slave protocol, where the gateway is the slave servicing commands from
the master, which is the call agent. In the Cisco AVVID environment, the
CallManager functions as the call agent.Two benefits MGCP provides over
H.323 are centralized dial plans and dynamic dial plan updates versus statically
configuring each H.323 gateway. (The MGCP protocol communicates over UDP
port 2427 and TCP port 2428.)
AVVID Gateway Selection • Chapter 3 67
68 Chapter 3 • AVVID Gateway Selection
The skinny gateways are the DT-24+, DE-30+, and Catalyst 4000/6000
modules, which provide CallManager access to digital gateways. An example of
an H.323 gateway is a Cisco IOS router like the 2600 and 3600.The VG-200 is
an MGCP gateway with future support for the 2600, 3600, 3810, and Catalyst
modules.
Another protocol being implemented in Cisco gateways is the Session
Initiation Protocol. SIP (described in RFC 2543) is an application-layer control
protocol that can establish, modify, and terminate multimedia sessions or calls.
These sessions include IP conferences, telephone calls, and multimedia distribution.
A Cisco VoIP solution for SIP consists of a SIP agent, 7960 IP Phone, SIP
gateway, and a SIP proxy server.
SIP supports five elements of establishing and terminating communications:
User location
User capabilities
User availability
Call setup
Call handling
Currently, the VoIP world is dominated by H.323; the emergence of SIP and
the increasing number of applications supporting this new technology means the
interoperability of SIP with existing H.323 networks. SIP was first available in
IOS version 12.1(1)T. SIP was enhanced in IOS version 12.2(2)XA, supporting
the Cisco 2600 and 3600 Series router-voice gateways, and the Cisco AS5300,
AS5350, and AS5400 access server-voice gateways now allow both SIP and H.323
to fully coexist, as well as allowing interoperability between the two protocols.
NOTE
An example of new software utilizing the functionality of SIP is the application
Windows Messenger, which is part of Windows XP. Windows
Messenger is real-time communications software that provides end-toend
IP telephony. SIP is a multipart communication protocol, but the first
version of Windows Messenger will only support a two-way conversation.
Video gateways are used to convert form H.320 devices to H.323 devices.
The Cisco IP/VC products allow companies to utilize their legacy H.320 ISDNwww
Internet Protocol (IP) telephony-PSTN-MGCP-SSP
Internet Protocol (IP) telephony and IP-based video conferencing solutions present
many opportunities to your organization, and at the same time introduce an
entirely new set of challenges to overcome. IP telephony benefits your organization
by providing simplified administration, toll bypass, and a unified messaging
platform. All of these benefits have the potential to save your organization a great
deal both administratively and monetarily when implemented correctly.
Several new IP telephony-specific enhancements need to be made to your
infrastructure in order to make IP telephony a reality.The CallManager system
replaces the traditional Private Branch Exchange (PBX) system for call processing;
currently in release 3.1, it has the potential to support up to 2500 IP telephones
per server and 10,000 per cluster. IP telephone handsets provide the user
interface to the IP telephony infrastructure. Currently in their second-generation,
Cisco offers four IP telephone handsets: the 7910/7910+SW, 7940, 7960, and
7935. Gateways provide the interface to the public switched telephone network
(PSTN). Cisco products currently support three protocols: the Skinny Station
Protocol (SSP), H.323, and Media Gateway Control Protocol (MGCP).The
Cisco Unity product suite offers a unified messaging solution, integrating voice,
video and fax communication into one medium.
IP telephony applications are used to enhance the IP telephony product
offering. Cisco has developed several IP telephony applications to work within
the AVVID product offering.WebAttendant is a software-based attendant console
used to replace the traditional PBX attendant console. It provides call monitoring
and management functions and can be used to monitor up to 26 conversations
concurrently. IP SoftPhone is a software-based version of an IP telephone, providing
a viable alternative for traveling users or others who do not have access to
an IP telephone. Internet Communications Software (ICS) is a grouping of five
tools: Automatic Call Distribution (ACD), Cisco IP Contact Center (IPCC),
Intelligent Contact Management (ICM), the Customer Interaction Suite, and
Network Applications Manager (NAM), for service and application providers.
Interactive voice response (IVR) and AutoAttendant are used for menu-based
telephone systems. Other vendors have also developed applications to work with
Cisco’s IP telephony solutions, which includes Interactive Intelligence, Latitude,
and Intelligent Telemanagement Solutions (ISI). Interactive Intelligence offers the
Interaction Center platform, installed on the ICS 7750. Latitude offers the
MeetingPlace IP software for video-conferencing, and ISI offers an accounting
and billing application that utilizes the call detail record (CDR) of each call.
www.syngress.com
New World Technologies • Chapter 2 61
Video over IP (VoIP) is made possible through the use of several devices
including gateways, gatekeepers, multi-point control units (MCU), video terminal
adapters (VTA), and endpoints. Gateways provide access to the outside world
from your internal network. Gatekeepers are used to permit or deny requests for
video conferences. MCUs serve as a center for video-conferencing communications
and infrastructure.The VTA’s role in video conferencing is to provide an
interface to legacy video-conferencing systems. Endpoints are the end-user
devices that subscribe to and receive services from video-conferencing. Cisco
offers the IPVC family of products to meet the video-conferencing needs of
organizations. Cisco also offers the IP/TV family of products for one-way video
broadcasting.
Additional infrastructure, including switches and routers, will also need to be
adapted in order to meet the needs of AVVID. Cisco’s voice-ready routers include
the 1750 Series, 2600 Series, 3600 Series, and 7200 Series. All of Cisco’s switches
support IP telephony, but the 3524XL-PWR, 4000 Series, and 6000 Series support
inline power. Inline power can come in the form of inline power modules
for the Catalyst switches or external power-patch panels.
Queuing and class of service (COS) measures must also be taken on your
LAN/WAN in order to ensure real-time delivery of voice and video traffic.
802.1Q provides a Layer 2 queuing solution for your LAN environment.
Solutions Fast Track
Introduction to IP Telephony
Simplified administration is achieved by converging three separate
networks into one, allowing one resource pool to administer the entire
network.
Toll bypass allows organizations to avoid costly telecommunications
expenses by utilizing the data infrastructure.
Unified messaging combines voice-mail, e-mail, and faxes into one easyto-
use interface.
IP Telephony Components
CallManager provides the IP telephony network with a software-based
PBX system.
www.syngress.com
62 Chapter 2 • New World Technologies
IP telephones provide the user interface to the IP telephony network.
Gateways provide the interface between the IP telephony network and
the public switched telephone network (PSTN) or a legacy PBX device.
Exploring IP Telephony Applications
WebAttendant replaces the traditional PBX attendant console.
IP SoftPhone provides a software-based IP telephone handset.
Third-party applications include software from Interactive Intelligence,
Latitude, and ISI.
Introduction to Video
Traditional video-conferencing utilizes ISDN lines in a point-to-point
infrastructure.
IP-based video-conferencing utilizes the H.323 specification allowing
for video-conferencing over a variety of mediums.
IP-based video-conferencing is much more efficient than traditional
video-conferencing because the existing data infrastructure is utilized
opposed to a separate infrastructure.
Gateways provide access to the outside world from your internal
network.
Gatekeepers are used to permit or deny requests for video conferences.
Multi-point control units (MCU) serve as a center for videoconferencing
communications and infrastructure.
Enhancing Network Infrastructure
Routers provide gateway services and voice aggregation for IP telephony
by use of analog ports, FXO, FXS, E&M as well as digital trunking cards.
Routers that support IP telephony include the 1751, 2600 Series, 3600
Series, and 7200 Series.
www.syngress.com
New World Technologies • Chapter 2 63
Switches that support inline power modules include the 3524XL-PWR,
6000 Series, and 4000 Series.
Inline power is also provided by using the Catalyst inline power patch
panel.
What Does the Future Hold ?
Future revisions on CallManager include a call center solution.
Pizza box and integrated access devices will provide all-in-one
functionality for branch offices.
IOS-based versions of CallManager will further develop.
many opportunities to your organization, and at the same time introduce an
entirely new set of challenges to overcome. IP telephony benefits your organization
by providing simplified administration, toll bypass, and a unified messaging
platform. All of these benefits have the potential to save your organization a great
deal both administratively and monetarily when implemented correctly.
Several new IP telephony-specific enhancements need to be made to your
infrastructure in order to make IP telephony a reality.The CallManager system
replaces the traditional Private Branch Exchange (PBX) system for call processing;
currently in release 3.1, it has the potential to support up to 2500 IP telephones
per server and 10,000 per cluster. IP telephone handsets provide the user
interface to the IP telephony infrastructure. Currently in their second-generation,
Cisco offers four IP telephone handsets: the 7910/7910+SW, 7940, 7960, and
7935. Gateways provide the interface to the public switched telephone network
(PSTN). Cisco products currently support three protocols: the Skinny Station
Protocol (SSP), H.323, and Media Gateway Control Protocol (MGCP).The
Cisco Unity product suite offers a unified messaging solution, integrating voice,
video and fax communication into one medium.
IP telephony applications are used to enhance the IP telephony product
offering. Cisco has developed several IP telephony applications to work within
the AVVID product offering.WebAttendant is a software-based attendant console
used to replace the traditional PBX attendant console. It provides call monitoring
and management functions and can be used to monitor up to 26 conversations
concurrently. IP SoftPhone is a software-based version of an IP telephone, providing
a viable alternative for traveling users or others who do not have access to
an IP telephone. Internet Communications Software (ICS) is a grouping of five
tools: Automatic Call Distribution (ACD), Cisco IP Contact Center (IPCC),
Intelligent Contact Management (ICM), the Customer Interaction Suite, and
Network Applications Manager (NAM), for service and application providers.
Interactive voice response (IVR) and AutoAttendant are used for menu-based
telephone systems. Other vendors have also developed applications to work with
Cisco’s IP telephony solutions, which includes Interactive Intelligence, Latitude,
and Intelligent Telemanagement Solutions (ISI). Interactive Intelligence offers the
Interaction Center platform, installed on the ICS 7750. Latitude offers the
MeetingPlace IP software for video-conferencing, and ISI offers an accounting
and billing application that utilizes the call detail record (CDR) of each call.
www.syngress.com
New World Technologies • Chapter 2 61
Video over IP (VoIP) is made possible through the use of several devices
including gateways, gatekeepers, multi-point control units (MCU), video terminal
adapters (VTA), and endpoints. Gateways provide access to the outside world
from your internal network. Gatekeepers are used to permit or deny requests for
video conferences. MCUs serve as a center for video-conferencing communications
and infrastructure.The VTA’s role in video conferencing is to provide an
interface to legacy video-conferencing systems. Endpoints are the end-user
devices that subscribe to and receive services from video-conferencing. Cisco
offers the IPVC family of products to meet the video-conferencing needs of
organizations. Cisco also offers the IP/TV family of products for one-way video
broadcasting.
Additional infrastructure, including switches and routers, will also need to be
adapted in order to meet the needs of AVVID. Cisco’s voice-ready routers include
the 1750 Series, 2600 Series, 3600 Series, and 7200 Series. All of Cisco’s switches
support IP telephony, but the 3524XL-PWR, 4000 Series, and 6000 Series support
inline power. Inline power can come in the form of inline power modules
for the Catalyst switches or external power-patch panels.
Queuing and class of service (COS) measures must also be taken on your
LAN/WAN in order to ensure real-time delivery of voice and video traffic.
802.1Q provides a Layer 2 queuing solution for your LAN environment.
Solutions Fast Track
Introduction to IP Telephony
Simplified administration is achieved by converging three separate
networks into one, allowing one resource pool to administer the entire
network.
Toll bypass allows organizations to avoid costly telecommunications
expenses by utilizing the data infrastructure.
Unified messaging combines voice-mail, e-mail, and faxes into one easyto-
use interface.
IP Telephony Components
CallManager provides the IP telephony network with a software-based
PBX system.
www.syngress.com
62 Chapter 2 • New World Technologies
IP telephones provide the user interface to the IP telephony network.
Gateways provide the interface between the IP telephony network and
the public switched telephone network (PSTN) or a legacy PBX device.
Exploring IP Telephony Applications
WebAttendant replaces the traditional PBX attendant console.
IP SoftPhone provides a software-based IP telephone handset.
Third-party applications include software from Interactive Intelligence,
Latitude, and ISI.
Introduction to Video
Traditional video-conferencing utilizes ISDN lines in a point-to-point
infrastructure.
IP-based video-conferencing utilizes the H.323 specification allowing
for video-conferencing over a variety of mediums.
IP-based video-conferencing is much more efficient than traditional
video-conferencing because the existing data infrastructure is utilized
opposed to a separate infrastructure.
Gateways provide access to the outside world from your internal
network.
Gatekeepers are used to permit or deny requests for video conferences.
Multi-point control units (MCU) serve as a center for videoconferencing
communications and infrastructure.
Enhancing Network Infrastructure
Routers provide gateway services and voice aggregation for IP telephony
by use of analog ports, FXO, FXS, E&M as well as digital trunking cards.
Routers that support IP telephony include the 1751, 2600 Series, 3600
Series, and 7200 Series.
www.syngress.com
New World Technologies • Chapter 2 63
Switches that support inline power modules include the 3524XL-PWR,
6000 Series, and 4000 Series.
Inline power is also provided by using the Catalyst inline power patch
panel.
What Does the Future Hold ?
Future revisions on CallManager include a call center solution.
Pizza box and integrated access devices will provide all-in-one
functionality for branch offices.
IOS-based versions of CallManager will further develop.
IP Telephony Gateways and Protocols
IP Telephony Gateways and Protocols
Gateway H.323 MGCP Skinny
Catalyst 4000 Access Yes (PSTN) Future Yes (Conferencing and
Gateway Module MTP transcoding services)
Catalyst 6000 Voice No Future Yes
T1/E1 Module
Cisco 1750 Yes Yes No
Cisco 2600 Yes Yes No
Cisco 3600 Yes Yes No
Cisco 3810 Yes Future No
Cisco 7200 Yes No No
Cisco 7500 Future No No
Cisco AS5300 Yes No No
Cisco DT-24+ and DE-30+ No No Yes
Cisco VG-200 Yes Yes No
www.syngress.com
AVVID Gateway Selection • Chapter 3 73
In the following sections, we will discuss the different hardware platforms in
regards to what interfaces and protocols are supported as well as what application
or environment is suitable.
Gateway H.323 MGCP Skinny
Catalyst 4000 Access Yes (PSTN) Future Yes (Conferencing and
Gateway Module MTP transcoding services)
Catalyst 6000 Voice No Future Yes
T1/E1 Module
Cisco 1750 Yes Yes No
Cisco 2600 Yes Yes No
Cisco 3600 Yes Yes No
Cisco 3810 Yes Future No
Cisco 7200 Yes No No
Cisco 7500 Future No No
Cisco AS5300 Yes No No
Cisco DT-24+ and DE-30+ No No Yes
Cisco VG-200 Yes Yes No
www.syngress.com
AVVID Gateway Selection • Chapter 3 73
In the following sections, we will discuss the different hardware platforms in
regards to what interfaces and protocols are supported as well as what application
or environment is suitable.
CallManager Redundancy -DTMF Relay -Supplemental Services
In choosing a gateway, attention should be given to ensure it supports three
important features:
CallManager Redundancy
DTMF Relay
Supplemental Services
Whether the installation is a large enterprise deployment with multiple
CallManager clusters or a simple two-node cluster, it’s important CallManager
Redundancy is supported by the gateway of choice. CallManager Redundancy is
required since an AVVID network needs to have the same high level of availability
as the traditional PBX.You do not want your access to a PBX or PSTN
compromised by a single point of failure. DTMF uses two frequencies, a high and
low tone to distinguish numbers on a telephone keypad.This signaling is usually
carried over a 64 Kbps voice circuit, and accomplished with little problem, but
with lower-bit CODEC the signal can be lost or unrecognizable.The gateways
provide out-of-band support for passing DTMF signals across the VoIP network
via the gateway protocols, as illustrated in Table 3.4.The AVVID gateway needs to
provide support for other user telephony services such as Call Hold, Call
Handling, and N-way Conference.These are normal traditional voice services,
which should be considered basic requirements for an AVVID network.
important features:
CallManager Redundancy
DTMF Relay
Supplemental Services
Whether the installation is a large enterprise deployment with multiple
CallManager clusters or a simple two-node cluster, it’s important CallManager
Redundancy is supported by the gateway of choice. CallManager Redundancy is
required since an AVVID network needs to have the same high level of availability
as the traditional PBX.You do not want your access to a PBX or PSTN
compromised by a single point of failure. DTMF uses two frequencies, a high and
low tone to distinguish numbers on a telephone keypad.This signaling is usually
carried over a 64 Kbps voice circuit, and accomplished with little problem, but
with lower-bit CODEC the signal can be lost or unrecognizable.The gateways
provide out-of-band support for passing DTMF signals across the VoIP network
via the gateway protocols, as illustrated in Table 3.4.The AVVID gateway needs to
provide support for other user telephony services such as Call Hold, Call
Handling, and N-way Conference.These are normal traditional voice services,
which should be considered basic requirements for an AVVID network.
Analog VoIP Gateways
Analog VoIP Gateways
Gateway E&M FXO FXS DID/CLID
Catalyst 4000 Access Yes Yes Yes 12.1(5)T/12.1(5)T
Gateway Module
Catalyst 6000 Voice No No Yes No/Yes
T1/E1 Module
Cisco 1750 Yes Yes Yes Future
Cisco 2600 Yes Yes Yes 12.1(3)T/12.1(2)XH
Cisco 3600 Yes Yes Yes 12.1(3)T/12.1(2)XH
Cisco 3810 Yes Yes Yes 12.1(3)T/12.1(2)XH
Cisco AS5300 No No No N/A
Cisco 7200 No No No N/A
Cisco 7500 No No No N/A
Cisco DT-24+ and DE-30+ No No No N/A
Cisco VG-200 H.323v2 Yes Yes 12.1(5)XM1
If higher capacity voice channels are required to either the PSTN or PBX, a
digital gateway may be more effective.Table 3.3 lists the interfaces and features
supported on the various hardware platforms. The different gateways support two
main signaling types: either ISDN Primary Rate Interface (PRI) or channel associated
signaling (CAS) for T1 or E1. ISDN PRI, meanwhile, utilizes a “D”
channel for signaling. ISDN PRI is classified as out-of-band signaling since there
is a channel dedicated for signaling, whereas, CAS signaling (also referred to as
robbed-bit signaling) uses some of the bandwidth from each channel. CAS signaling
forms include loop start, ground start, and E&M.T1CAS supports automatic
number identifier (ANI) and dialed number identification service (DNIS)
as well, which are also known as Caller ID and Called Party Number. DNIS
returns to the called number they dialed. Determining which PRI type of interface
is required depends on whether you’re connecting your gateway to a PBX
or PSTN.Typically, if the gateway is connecting to a PBX, you will need a
Network Side PRI interface, since the PBX is on the “user side.” Normally, the
PSTN (with a switch like a DMS100) functions as the “network side” and the
gateway needs a User Side PRI interface.The gateway could be connected to
both the PSTN and PBX with ISDN PRI configured appropriately for each
interface.
www.syngress.com
Gateway E&M FXO FXS DID/CLID
Catalyst 4000 Access Yes Yes Yes 12.1(5)T/12.1(5)T
Gateway Module
Catalyst 6000 Voice No No Yes No/Yes
T1/E1 Module
Cisco 1750 Yes Yes Yes Future
Cisco 2600 Yes Yes Yes 12.1(3)T/12.1(2)XH
Cisco 3600 Yes Yes Yes 12.1(3)T/12.1(2)XH
Cisco 3810 Yes Yes Yes 12.1(3)T/12.1(2)XH
Cisco AS5300 No No No N/A
Cisco 7200 No No No N/A
Cisco 7500 No No No N/A
Cisco DT-24+ and DE-30+ No No No N/A
Cisco VG-200 H.323v2 Yes Yes 12.1(5)XM1
If higher capacity voice channels are required to either the PSTN or PBX, a
digital gateway may be more effective.Table 3.3 lists the interfaces and features
supported on the various hardware platforms. The different gateways support two
main signaling types: either ISDN Primary Rate Interface (PRI) or channel associated
signaling (CAS) for T1 or E1. ISDN PRI, meanwhile, utilizes a “D”
channel for signaling. ISDN PRI is classified as out-of-band signaling since there
is a channel dedicated for signaling, whereas, CAS signaling (also referred to as
robbed-bit signaling) uses some of the bandwidth from each channel. CAS signaling
forms include loop start, ground start, and E&M.T1CAS supports automatic
number identifier (ANI) and dialed number identification service (DNIS)
as well, which are also known as Caller ID and Called Party Number. DNIS
returns to the called number they dialed. Determining which PRI type of interface
is required depends on whether you’re connecting your gateway to a PBX
or PSTN.Typically, if the gateway is connecting to a PBX, you will need a
Network Side PRI interface, since the PBX is on the “user side.” Normally, the
PSTN (with a switch like a DMS100) functions as the “network side” and the
gateway needs a User Side PRI interface.The gateway could be connected to
both the PSTN and PBX with ISDN PRI configured appropriately for each
interface.
www.syngress.com
Choosing a Voice Gateway Solution
Choosing a Voice Gateway Solution
There are a number of different voice gateways available for CallManager and
VoIP implementations, which are divided into categories by type of gateway and
the protocol running for gateway communications.The gateway selection is based
on some of the following variables: analog or digital, capacity, connection type,
services, features, and country of installation.Table 3.2 lists the analog VoIP gateways
and the respective voice interface cards (VICs) supported.The analog gateways
provide connectivity to analog phone sets, central office, and PBX.The
Foreign Exchange Station (FXS) ports are used to provide dial tone for analog
phones, faxes, and speakerphones, while a Foreign Exchange Office (FXO) port
in a gateway is for connectivity to Central Office for analog access to the PSTN.
Ear-and-mouth (E&M) ports, on the other hand, are for PBX-to-PBX signaling
communication.You can determine what type of analog VoIP gateway you need
by answering one of the following questions:Are FXO ports required for PSTN
connectivity or is Direct Inward Dial (DID) a necessity? Another factor in
selecting an analog gateway may be capacity. For example, if you require a large
number of FXS ports for legacy analog connections, you would select a Cisco
3660 or a Catalyst 6000 with a 24-port FXS module rather than a smaller
capacity gateway such as a VG-200 or Cisco 2600.
There are a number of different voice gateways available for CallManager and
VoIP implementations, which are divided into categories by type of gateway and
the protocol running for gateway communications.The gateway selection is based
on some of the following variables: analog or digital, capacity, connection type,
services, features, and country of installation.Table 3.2 lists the analog VoIP gateways
and the respective voice interface cards (VICs) supported.The analog gateways
provide connectivity to analog phone sets, central office, and PBX.The
Foreign Exchange Station (FXS) ports are used to provide dial tone for analog
phones, faxes, and speakerphones, while a Foreign Exchange Office (FXO) port
in a gateway is for connectivity to Central Office for analog access to the PSTN.
Ear-and-mouth (E&M) ports, on the other hand, are for PBX-to-PBX signaling
communication.You can determine what type of analog VoIP gateway you need
by answering one of the following questions:Are FXO ports required for PSTN
connectivity or is Direct Inward Dial (DID) a necessity? Another factor in
selecting an analog gateway may be capacity. For example, if you require a large
number of FXS ports for legacy analog connections, you would select a Cisco
3660 or a Catalyst 6000 with a 24-port FXS module rather than a smaller
capacity gateway such as a VG-200 or Cisco 2600.
Video Image Format Standards
based videoconferencing to integrate with newer IP-based H.323 videoconferencing
devices. Cisco IP/VC gateways support H.261 and H.263 video coders/
decoders (CODECs); H.261 is used as multiple channels of 64 Kbps, while
H.263 is a higher quality video CODEC.Table 3.1 lists the formats and image
sizes supported by H.261 and H.263.
Table 3.1 Video Image Format Standards
Format Image Size H.261 H.263
Sub-QCIF 128x96 Optional Required
QCIF 176x144 Required Required
CIF 352x288 Optional Optional
4CIF 702x576 N/A Optional
16CIF 1408x1152 N/A Optional
devices. Cisco IP/VC gateways support H.261 and H.263 video coders/
decoders (CODECs); H.261 is used as multiple channels of 64 Kbps, while
H.263 is a higher quality video CODEC.Table 3.1 lists the formats and image
sizes supported by H.261 and H.263.
Table 3.1 Video Image Format Standards
Format Image Size H.261 H.263
Sub-QCIF 128x96 Optional Required
QCIF 176x144 Required Required
CIF 352x288 Optional Optional
4CIF 702x576 N/A Optional
16CIF 1408x1152 N/A Optional
What Does the Future Hold-AVVID
What Does the Future Hold?
When we discuss the future of AVVID, it is useful to examine where we are and
how we got here. Over the past few years, this technology has grown by leaps
and bounds.Advances have been made in every area, with new offerings coming
out almost every day.
An area that will definitely be enhanced further will be the CallManager
platform. In version 2.x organizations were limited to 200 IP telephones per
server.When 3.0 debuted, this number went up to 2500 with support for 10,000
IP phones within a cluster.With the latest release, 3.1, we can now support up to
1,000,000 IP telephones. I would expect that this number will continue to rise
with the later revisions of the software. As this technology gains further acceptance,
we can also probably expect to see several enhancements to the
CallManager feature offering, such as a voice-recording system. I would also
expect to see support specifically for call centers. IOS-based versions of the
CallManager platform are already available; although these currently provide a
very limited feature set, future revisions will most likely contain more features
and open IP telephony solutions up for branch offices.
Aside from CallManager, I would also expect that we will see further software
developments and new packages from both Cisco and other vendors. A good
example would be the Intelligent Contact Management (ICM) suite, due to be
released in the first part of 2002. It is a software solution used for the direction
and relay of customer contact information between resources.This system will
utilize a set of user-defined roles in order to route voice,WWW, and e-mail correspondence
to the appropriate system or resource.This and other systems of its
kind will further enhance and augment the IP telephony and AVVID solutions.
Another area we will likely see growth is with the offering of pizza box
solutions.These products derive their colorful name from their small form-factor,
which is about the size of a pizza box.These are integrated all-in-one access solutions
that provide capabilities such as routing, switching, and voice-gateway services.
A good example of this type of solution would be the IAD1101 integrated
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SONET, or Gigabit Ethernet. Products found at this layer are
typically high-end routers such as the 7500, or Layer 3
switching devices like the 12000 Gigabit Switch Router (GSR).
Concerning IP telephony, we would probably not see much
here, as it should be implemented at the lower levels.
New World Technologies • Chapter 2 59
access device.This solution offers two T1 ports, a v.35 port, a 10bastT Ethernet
port and an RS-232 port. Eight analog ports are also available via an expansion
slot.This type of solution offers several capabilities of the larger scale platforms at
a fraction of the cost.
At the client end, I would expect to see a third generation of IP telephones
offering wider support for the SIP protocol as well as other features, and maybe
eventually integrating with a video endpoint device, although Cisco has not
made any indication of this.
Without our standard-issue crystal ball, we cannot be certain as to what the
future holds, but if past performance is any indication, I would expect we will
continue to see rapid development of exciting products and services. Perhaps IP
telephony will one day replace our existing telephone infrastructure. Only time
will tell.
When we discuss the future of AVVID, it is useful to examine where we are and
how we got here. Over the past few years, this technology has grown by leaps
and bounds.Advances have been made in every area, with new offerings coming
out almost every day.
An area that will definitely be enhanced further will be the CallManager
platform. In version 2.x organizations were limited to 200 IP telephones per
server.When 3.0 debuted, this number went up to 2500 with support for 10,000
IP phones within a cluster.With the latest release, 3.1, we can now support up to
1,000,000 IP telephones. I would expect that this number will continue to rise
with the later revisions of the software. As this technology gains further acceptance,
we can also probably expect to see several enhancements to the
CallManager feature offering, such as a voice-recording system. I would also
expect to see support specifically for call centers. IOS-based versions of the
CallManager platform are already available; although these currently provide a
very limited feature set, future revisions will most likely contain more features
and open IP telephony solutions up for branch offices.
Aside from CallManager, I would also expect that we will see further software
developments and new packages from both Cisco and other vendors. A good
example would be the Intelligent Contact Management (ICM) suite, due to be
released in the first part of 2002. It is a software solution used for the direction
and relay of customer contact information between resources.This system will
utilize a set of user-defined roles in order to route voice,WWW, and e-mail correspondence
to the appropriate system or resource.This and other systems of its
kind will further enhance and augment the IP telephony and AVVID solutions.
Another area we will likely see growth is with the offering of pizza box
solutions.These products derive their colorful name from their small form-factor,
which is about the size of a pizza box.These are integrated all-in-one access solutions
that provide capabilities such as routing, switching, and voice-gateway services.
A good example of this type of solution would be the IAD1101 integrated
www.syngress.com
SONET, or Gigabit Ethernet. Products found at this layer are
typically high-end routers such as the 7500, or Layer 3
switching devices like the 12000 Gigabit Switch Router (GSR).
Concerning IP telephony, we would probably not see much
here, as it should be implemented at the lower levels.
New World Technologies • Chapter 2 59
access device.This solution offers two T1 ports, a v.35 port, a 10bastT Ethernet
port and an RS-232 port. Eight analog ports are also available via an expansion
slot.This type of solution offers several capabilities of the larger scale platforms at
a fraction of the cost.
At the client end, I would expect to see a third generation of IP telephones
offering wider support for the SIP protocol as well as other features, and maybe
eventually integrating with a video endpoint device, although Cisco has not
made any indication of this.
Without our standard-issue crystal ball, we cannot be certain as to what the
future holds, but if past performance is any indication, I would expect we will
continue to see rapid development of exciting products and services. Perhaps IP
telephony will one day replace our existing telephone infrastructure. Only time
will tell.
6000 Series Switches
6000 Series Switches
The 6000 Series is a highly scalable, enterprise class series of switches.The 6000
Series offers a completely modular design, utilizing supervisor modules, with the
capability for redundant supervisor modules, if necessary.There are five switches
in the 6000 Series family: the 6006, 6009, 6506, 6509, and 6513.The 6006 and
6506 offer six slots while the 6009 and 6509 offer nine slots.The 6513 is the
largest form-factor in the product line, offering 13 slots.The 6000 Series provides
inline power directly through the use of specialized 48-port switching blades.The
6006 and 6506 can support up to 240 10/100 ports, while the 6009 and 6509
can support up to 384 10/100 ports, and the 6513 can support up to 576 10/100
ports. Gateway functionality is provided via the WS-X6608-x1 module.This
module supports SSP and will in the future support MGCP.The 6000 Series also
offers an eight-port voice T1/E1 and services module to provide connectivity to
legacy PSTN or PBX systems, as well as a 24-port FXS module for analog telephone
connectivity.
The 6000 Series is a highly scalable, enterprise class series of switches.The 6000
Series offers a completely modular design, utilizing supervisor modules, with the
capability for redundant supervisor modules, if necessary.There are five switches
in the 6000 Series family: the 6006, 6009, 6506, 6509, and 6513.The 6006 and
6506 offer six slots while the 6009 and 6509 offer nine slots.The 6513 is the
largest form-factor in the product line, offering 13 slots.The 6000 Series provides
inline power directly through the use of specialized 48-port switching blades.The
6006 and 6506 can support up to 240 10/100 ports, while the 6009 and 6509
can support up to 384 10/100 ports, and the 6513 can support up to 576 10/100
ports. Gateway functionality is provided via the WS-X6608-x1 module.This
module supports SSP and will in the future support MGCP.The 6000 Series also
offers an eight-port voice T1/E1 and services module to provide connectivity to
legacy PSTN or PBX systems, as well as a 24-port FXS module for analog telephone
connectivity.
Exploring Inline Power Options
Exploring Inline Power Options
During our discussion earlier in this chapter concerning IP telephones, we discussed
how second-generation phones were superior to their first-generation
counterparts because they offered support for inline power. First-generation telephones
were limited in that they required an external power source in order to
function. Inline power allows second-generation phones to avoid this pitfall.
There are two ways in which inline power can be offered to second-generation
telephones, either by way of a power patch panel, or through the use of inline
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New World Technologies • Chapter 2 55
power modules installed directly in the switch. Let’s discuss these different power
options as well as their advantages and disadvantages.
During our discussion earlier in this chapter concerning IP telephones, we discussed
how second-generation phones were superior to their first-generation
counterparts because they offered support for inline power. First-generation telephones
were limited in that they required an external power source in order to
function. Inline power allows second-generation phones to avoid this pitfall.
There are two ways in which inline power can be offered to second-generation
telephones, either by way of a power patch panel, or through the use of inline
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New World Technologies • Chapter 2 55
power modules installed directly in the switch. Let’s discuss these different power
options as well as their advantages and disadvantages.
Different Queuing for Video/Voice
Different Queuing for Video/Voice
Queuing is an important design and performance issue that must also be examined
when discussing IP telephony. Queuing has traditionally been a Layer 3
function for WAN connections, but when discussing a converged network, specifically
that dealing with voice or video traffic, attention must also be given to the
LAN. Layer 2 traffic can be classified by type of service using the 802.1Q protocol.
It is recommended that when using this protocol you separate voice and
video traffic from regular data traffic and place this traffic in a higher-priority
queue. 802.1Q specifies seven classes of service (COS), 0 being lowest priority
and 7 being of the highest priority. It is recommended that COS 4–7 be used for
voice and video, and that 0–3 be used for normal data operations. An important
note to make regarding Layer 2 queuing is that once the packet encounters a
router, the Layer 2 information is lost—in other words, 802.1Q is only a LAN
solution. For traffic crossing WAN links, Layer 3 queuing must be incorporated.
Queuing is an important design and performance issue that must also be examined
when discussing IP telephony. Queuing has traditionally been a Layer 3
function for WAN connections, but when discussing a converged network, specifically
that dealing with voice or video traffic, attention must also be given to the
LAN. Layer 2 traffic can be classified by type of service using the 802.1Q protocol.
It is recommended that when using this protocol you separate voice and
video traffic from regular data traffic and place this traffic in a higher-priority
queue. 802.1Q specifies seven classes of service (COS), 0 being lowest priority
and 7 being of the highest priority. It is recommended that COS 4–7 be used for
voice and video, and that 0–3 be used for normal data operations. An important
note to make regarding Layer 2 queuing is that once the packet encounters a
router, the Layer 2 information is lost—in other words, 802.1Q is only a LAN
solution. For traffic crossing WAN links, Layer 3 queuing must be incorporated.
Power Cube
Power Cube
Power cubes are an external power supply, used as a sole means of power for the
first-generation telephone offerings. Power cubes can be used by second-generation
telephones as a sole means of power, or more commonly, can be used as a
backup power supply to the inline power patch panel and the inline power modules.
The advantage to this solution is that, when used with inline power, it provides
a redundant power supply for your IP telephone.When used solely as a
means of power, its advantage is that you can deploy IP telephones and not have
to replace your switches or install inline power patch panels for power.The major
disadvantage is that you must provide a power outlet for each cube, and it adds to
the mass of cables around a user’s desk
Power cubes are an external power supply, used as a sole means of power for the
first-generation telephone offerings. Power cubes can be used by second-generation
telephones as a sole means of power, or more commonly, can be used as a
backup power supply to the inline power patch panel and the inline power modules.
The advantage to this solution is that, when used with inline power, it provides
a redundant power supply for your IP telephone.When used solely as a
means of power, its advantage is that you can deploy IP telephones and not have
to replace your switches or install inline power patch panels for power.The major
disadvantage is that you must provide a power outlet for each cube, and it adds to
the mass of cables around a user’s desk
Power Patch Panel
Power Patch Panel
The Catalyst inline power patch panel offers an alternative to the forklift upgrade
that might be necessary in order to accommodate inline power.This solution
allows you to utilize your existing switching infrastructure, such as 2900 and 5000
Series Catalyst switches, by providing inline power external to the switch.The
Catalyst inline power patch panel offers 96 ports, for support of up to 48 stations
per panel, one port for the IP telephone and one port for the switch.The major
advantage to this solution is that it helps to protect your existing investment in
switches and helps to keep your options open to future product offerings.The
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56 Chapter 2 • New World Technologies
major disadvantage is that you now have an additional piece of equipment in
order to administer.
The Catalyst inline power patch panel offers an alternative to the forklift upgrade
that might be necessary in order to accommodate inline power.This solution
allows you to utilize your existing switching infrastructure, such as 2900 and 5000
Series Catalyst switches, by providing inline power external to the switch.The
Catalyst inline power patch panel offers 96 ports, for support of up to 48 stations
per panel, one port for the IP telephone and one port for the switch.The major
advantage to this solution is that it helps to protect your existing investment in
switches and helps to keep your options open to future product offerings.The
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56 Chapter 2 • New World Technologies
major disadvantage is that you now have an additional piece of equipment in
order to administer.
Inline Power Modules
Inline Power Modules
Inline power is currently available for three switches in the Catalyst product line:
the 3524XL-PWR, the 4000 Series, and the 6000 Series.The 3524XL-PWR is a
fixed-configuration 24-port switch. It provides out-of-the-box inline power support.
An important note to make is that the 3524XL-PWR switch offers no
inline power redundancy.
The 4000 Series provides inline power through use of the Catalyst 4000
Inline power 10/100BaseT Switching module and the Power Entry Module
(PEM). Redundancy is provided to the 4006 by use of the WS-P4603 auxiliary
DC power shelf.This allows for an N+1 protection scheme protecting against a
single power supply failure. An important note to make is that the 4003 cannot
interface with the power entry module and therefore cannot utilize inline power
directly from the switch. In order for the 4003 to provide inline power, you must
use the Catalyst inline power patch panel.
The 6000 Series provides inline power by use of a 48-port switching blade.
Inline power is provided to the switch via 2500-watt power supply.Two power
supplies can be used for redundancy. Inline power modules offer a great solution
in environments where space is at a premium.
Because all of the functions are collapsed into one piece of equipment,
administration is simplified. On the down side, this solution may require a forklift
upgrade, as inline power modules are only available for the 3500, 4000, and 6000
Series, which could introduce a great deal of added expense. Even though powerredundancy
is available for these switches, you are still relying on a single point of
failure should the entire switch fail.
Inline power is currently available for three switches in the Catalyst product line:
the 3524XL-PWR, the 4000 Series, and the 6000 Series.The 3524XL-PWR is a
fixed-configuration 24-port switch. It provides out-of-the-box inline power support.
An important note to make is that the 3524XL-PWR switch offers no
inline power redundancy.
The 4000 Series provides inline power through use of the Catalyst 4000
Inline power 10/100BaseT Switching module and the Power Entry Module
(PEM). Redundancy is provided to the 4006 by use of the WS-P4603 auxiliary
DC power shelf.This allows for an N+1 protection scheme protecting against a
single power supply failure. An important note to make is that the 4003 cannot
interface with the power entry module and therefore cannot utilize inline power
directly from the switch. In order for the 4003 to provide inline power, you must
use the Catalyst inline power patch panel.
The 6000 Series provides inline power by use of a 48-port switching blade.
Inline power is provided to the switch via 2500-watt power supply.Two power
supplies can be used for redundancy. Inline power modules offer a great solution
in environments where space is at a premium.
Because all of the functions are collapsed into one piece of equipment,
administration is simplified. On the down side, this solution may require a forklift
upgrade, as inline power modules are only available for the 3500, 4000, and 6000
Series, which could introduce a great deal of added expense. Even though powerredundancy
is available for these switches, you are still relying on a single point of
failure should the entire switch fail.
4000 Series Switches
4000 Series Switches
The 4000 Series is a step up from the 3500, offering a modular configuration in
four different switches: the 4003, 4006, 4840G, and 4908G.The 4000 Series also
offers supervisor engine functionality, similar to that of the 5500 Series.Within
the 4000 Series, the 4006 is currently the only switch to offer inline power; by
use of the Catalyst 4000 inline power 10/100BaseT switching module or the use
of an auxiliary power shelf, the 4006 supports up to 240 10/100 ports.The 4003
is also an Ethernet switch, very similar to the 4006, but unfortunately the 4003
cannot offer inline power functionality.The 4840G and 4908G are both gigabit
Ethernet switches.The 4000 Series also offers voice-gateway functionality
through the use of the Series 4000 WS-X4604-GWY module, which provides
support for both H.323 and SSP (in the future it will support MGCP).
The 4000 Series is a step up from the 3500, offering a modular configuration in
four different switches: the 4003, 4006, 4840G, and 4908G.The 4000 Series also
offers supervisor engine functionality, similar to that of the 5500 Series.Within
the 4000 Series, the 4006 is currently the only switch to offer inline power; by
use of the Catalyst 4000 inline power 10/100BaseT switching module or the use
of an auxiliary power shelf, the 4006 supports up to 240 10/100 ports.The 4003
is also an Ethernet switch, very similar to the 4006, but unfortunately the 4003
cannot offer inline power functionality.The 4840G and 4908G are both gigabit
Ethernet switches.The 4000 Series also offers voice-gateway functionality
through the use of the Series 4000 WS-X4604-GWY module, which provides
support for both H.323 and SSP (in the future it will support MGCP).
3500 Series Switches
3500 Series Switches
The 3500 Series is a scalable, entry-level solution for small- to mid-sized networks.
It is a wholly Cisco-developed switch, which runs a router-like IOS.The
3500 Series of switches are fixed configuration switches, all offering 10/100
Ethernet ports and Gigabit Interface Converter (GBIC) ports.The difference
comes in the number of ports offered and the forwarding rate at which the
switch can process packets. Currently the 3524XL-PWR is the only switch in
the 3500 Series that supports inline power, although other models within the
35xx product line will, in the future, most likely offer inline power as well.The
3524XL-PWR offers 24 10/100 Ethernet ports, 2 GBIC ports, and a packetforwarding
rate of 6.5 million packets per second.
The 3500 Series is a scalable, entry-level solution for small- to mid-sized networks.
It is a wholly Cisco-developed switch, which runs a router-like IOS.The
3500 Series of switches are fixed configuration switches, all offering 10/100
Ethernet ports and Gigabit Interface Converter (GBIC) ports.The difference
comes in the number of ports offered and the forwarding rate at which the
switch can process packets. Currently the 3524XL-PWR is the only switch in
the 3500 Series that supports inline power, although other models within the
35xx product line will, in the future, most likely offer inline power as well.The
3524XL-PWR offers 24 10/100 Ethernet ports, 2 GBIC ports, and a packetforwarding
rate of 6.5 million packets per second.
Cisco Switches
Cisco Switches
Cisco’s Catalyst switch line is a highly-advanced line of switching solutions that
scale to meet various business needs, from small organizations to multinational
corporations. Catalyst switches operate at Layer 2, but Layer 3 switching is also
possible with a Route Switch Module (RSM). All of the switches in the Catalyst
line support AVVID networks, including IP phones, but specific switches within
the product line are designed specifically to meet the needs of IP telephony,
specifically inline power, which we will discuss in the next section, and gateway
functionality.We will discuss three lines that meet this challenge: the 3500, 4000,
and 6000 Series.
Cisco’s Catalyst switch line is a highly-advanced line of switching solutions that
scale to meet various business needs, from small organizations to multinational
corporations. Catalyst switches operate at Layer 2, but Layer 3 switching is also
possible with a Route Switch Module (RSM). All of the switches in the Catalyst
line support AVVID networks, including IP phones, but specific switches within
the product line are designed specifically to meet the needs of IP telephony,
specifically inline power, which we will discuss in the next section, and gateway
functionality.We will discuss three lines that meet this challenge: the 3500, 4000,
and 6000 Series.
3600 Series Routers
3600 Series Routers
The 3600 Series bears many resemblances to the 2600 Series, but the 3600 Series is
quite a bit more powerful, offering a great deal more scalability and processing
functions.There are three classes in the 3600 Series: the 3620, 3640, and 3660.The
3620 provides two expansion module slots, the 3640 offers four, and the 3660
offers six.Whereas the 2600 supports the use of WICs, the 3600 Series supports the
use of carrier cards that provide service for WAN, LAN, and voice interfaces; these
cards are interchangeable with the 2600 and 1750 Series routers. LAN support for
the 3600 supports 10 and 100 Mbps Ethernet, as well as token ring.
The 3600 Series bears many resemblances to the 2600 Series, but the 3600 Series is
quite a bit more powerful, offering a great deal more scalability and processing
functions.There are three classes in the 3600 Series: the 3620, 3640, and 3660.The
3620 provides two expansion module slots, the 3640 offers four, and the 3660
offers six.Whereas the 2600 supports the use of WICs, the 3600 Series supports the
use of carrier cards that provide service for WAN, LAN, and voice interfaces; these
cards are interchangeable with the 2600 and 1750 Series routers. LAN support for
the 3600 supports 10 and 100 Mbps Ethernet, as well as token ring.
7200 Series Routers
7200 Series Routers
The 7200 Series is Cisco’s first-level enterprise router. It offers a four- or six-slot
configuration with interfaces including ATM, Synchronous Optical Network
Technologies (SONET), ISDN BRI, ISDN PRI,T1, E1,T3, and E3. It also supports
AVVID applications through use of the multiservice interchange (MIX)
functionality.The MIX allows the 7200 to support digital voice as well as
gateway functionality through the use of two different trunk interfaces, the highcapacity
and medium-capacity T1/E1 trunk interface cards.The primary difference
between the two cards is that the high-capacity card includes an on-board
DSP card for compression.The 7200 Series can support up to 120 voice calls
depending on the module configuration used.This router also supports analog
voice applications through the use of voice interface cards (VICs).
The 7200 Series is Cisco’s first-level enterprise router. It offers a four- or six-slot
configuration with interfaces including ATM, Synchronous Optical Network
Technologies (SONET), ISDN BRI, ISDN PRI,T1, E1,T3, and E3. It also supports
AVVID applications through use of the multiservice interchange (MIX)
functionality.The MIX allows the 7200 to support digital voice as well as
gateway functionality through the use of two different trunk interfaces, the highcapacity
and medium-capacity T1/E1 trunk interface cards.The primary difference
between the two cards is that the high-capacity card includes an on-board
DSP card for compression.The 7200 Series can support up to 120 voice calls
depending on the module configuration used.This router also supports analog
voice applications through the use of voice interface cards (VICs).
2600 Series Routers
2600 Series Routers
The 2600 Series of Cisco’s routers has become one of the most popular connectivity
solutions for branch office connectivity. It offers a modular design, sharing
network modules with the 1600, 1700, and 3600 Series of routers, providing two
WAN interface card (WIC) connections as well as one network module slot.The
2600 router supports 10 Mbps and 100 Mbps Ethernet interfaces as well as token
ring.The 2600 supports VoIP applications and support for up to 48 digital voice
lines (60 in Europe).Voice interface cards (VICs) allow the 2600 Series to support
analog voice interfaces. By using two VIC cards in the WIC card slots, the 2600
can support up to four analog lines.
The 2600 Series of Cisco’s routers has become one of the most popular connectivity
solutions for branch office connectivity. It offers a modular design, sharing
network modules with the 1600, 1700, and 3600 Series of routers, providing two
WAN interface card (WIC) connections as well as one network module slot.The
2600 router supports 10 Mbps and 100 Mbps Ethernet interfaces as well as token
ring.The 2600 supports VoIP applications and support for up to 48 digital voice
lines (60 in Europe).Voice interface cards (VICs) allow the 2600 Series to support
analog voice interfaces. By using two VIC cards in the WIC card slots, the 2600
can support up to four analog lines.
1750 Multi-Service Series Routers
1750 Multi-Service Series Routers
The 1700 Series of routers provide a small office solution for organizations. As a
member of the 1700 Series family, the 1750 multiservice router series also offers
an IP telephony solution, two analog voice channels, a DSP, and three network
interface module slots for additional voice/data support.The 1750 can share the
same WAN and voice interface cards as the 2600 Series.This router would most
likely serve the small and home office market, due to its small capacity and limited
features—it would not be adequate in the larger branch office role.
The 1700 Series of routers provide a small office solution for organizations. As a
member of the 1700 Series family, the 1750 multiservice router series also offers
an IP telephony solution, two analog voice channels, a DSP, and three network
interface module slots for additional voice/data support.The 1750 can share the
same WAN and voice interface cards as the 2600 Series.This router would most
likely serve the small and home office market, due to its small capacity and limited
features—it would not be adequate in the larger branch office role.
MC 3810 Router
MC 3810 Router
The Multi-Service Access Concentrator 3810 (MC 3810) represents the first
router of its type, offering the full capabilities of a router as well as Voice over
Frame Relay,ATM, and leased lines. It was designed to be an all-inclusive solution
for branch-office deployments. A major disadvantage of the MC 3810 is that
it is expensive and the network modules used in it are not interchangeable with
any other platforms.This is no longer a very popular platform due to the VoIP
capabilities of routers such as the 2600 and 3600.
The Multi-Service Access Concentrator 3810 (MC 3810) represents the first
router of its type, offering the full capabilities of a router as well as Voice over
Frame Relay,ATM, and leased lines. It was designed to be an all-inclusive solution
for branch-office deployments. A major disadvantage of the MC 3810 is that
it is expensive and the network modules used in it are not interchangeable with
any other platforms.This is no longer a very popular platform due to the VoIP
capabilities of routers such as the 2600 and 3600.
Digital Voice Interfaces
Digital Voice Interfaces
Digital voice interfaces are provided to Cisco routers by use of digital voice
trunking cards and Digital Voice Processor (DVP) voice compression modules
(VCMs). Digital voice trunking cards interface most commonly with ISDN BRI
and PRI lines. By utilizing the individual channels on each line, it allows for a
single line to support two voice lines using BRI and up to 23 lines using PRI in
the U.S., and up to 30 in Europe. Digital voice processor VCMs allow a router to
take a voice conversation and compress it down to as small as 5.3 Kbps,
depending on the method utilized, as opposed to a 56 Kbps channel.This allows
for a much greater utilization of available bandwidth.
Digital voice interfaces are provided to Cisco routers by use of digital voice
trunking cards and Digital Voice Processor (DVP) voice compression modules
(VCMs). Digital voice trunking cards interface most commonly with ISDN BRI
and PRI lines. By utilizing the individual channels on each line, it allows for a
single line to support two voice lines using BRI and up to 23 lines using PRI in
the U.S., and up to 30 in Europe. Digital voice processor VCMs allow a router to
take a voice conversation and compress it down to as small as 5.3 Kbps,
depending on the method utilized, as opposed to a 56 Kbps channel.This allows
for a much greater utilization of available bandwidth.
Ear-and-Mouth
Ear-and-Mouth
Ear-and-mouth (E&M) offers a more advanced solution than either the FXO or
FXS ports, as well as several features that the other two do not, such as trunking
and either analog or digital transmission. E&M utilizes an RJ-48 port as opposed
to the RJ-11 used by the others. Cisco routers would most likely use an E&M
port for connection to PBX or PSTN, as well as a connection requiring
trunking.
Ear-and-mouth (E&M) offers a more advanced solution than either the FXO or
FXS ports, as well as several features that the other two do not, such as trunking
and either analog or digital transmission. E&M utilizes an RJ-48 port as opposed
to the RJ-11 used by the others. Cisco routers would most likely use an E&M
port for connection to PBX or PSTN, as well as a connection requiring
trunking.
Foreign Exchange Office
Foreign Exchange Office
Foreign Exchange Office (FXO) ports also utilize a standard RJ-11 telephone
jack. FXS ports are commonly used by businesses to connect their legacy PBX
systems to the service provider’s telephone network. Cisco routers can use an
FXO port to connect to a legacy PBX device or to directly connect to the
PSTN.
Foreign Exchange Office (FXO) ports also utilize a standard RJ-11 telephone
jack. FXS ports are commonly used by businesses to connect their legacy PBX
systems to the service provider’s telephone network. Cisco routers can use an
FXO port to connect to a legacy PBX device or to directly connect to the
PSTN.
Foreign Exchange Station
Foreign Exchange Station
Foreign Exchange Station (FXS) ports use a standard RJ-11 telephone jack to
connect to telephone handsets, modems, or fax machines.This is the common
type of interface found in homes. Cisco routers would most likely use this interface
for phone-to-phone connectivity.
Foreign Exchange Station (FXS) ports use a standard RJ-11 telephone jack to
connect to telephone handsets, modems, or fax machines.This is the common
type of interface found in homes. Cisco routers would most likely use this interface
for phone-to-phone connectivity.
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