Cisco Unified Videoconferencing Solution Reference Network Design (SRND)
Cisco Unified Videoconferencing Solution Reference Network Design (SRND)
1
No
Yes
Found
1
Not Found
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Chapter 7 Call Routing
Routing Using the Border Element
Figure 7-6 Via-Zone Processing
Routing Using the Border Element
Border elements can be deployed in a single gatekeeper zone or in multiple zones, as described in the
following sections.
Calls with a Single Zone
Gatekeeper deployments of one local zone can have the border element inserted for all calls or for calls
to and from remote zones. Although inserting border elements for all calls can have advantages, most
scenarios that need border element are for isolating calls to and from remote zones.
Figure 7-7 has a regional location with the gatekeeper and the border element. The local sites register
the video endpoints to the gatekeeper and use gatekeeper bandwidth for call admission control. The
border element is inserted into the calls for calls exiting the region to the IP PSTN, represented in the
gatekeeper configuration as the remote zone. (Example 7-2 lists the configuration details.)
2
5
0
4
1
4
Inputs:
Source endpoint
Source zone
Target zone
No Yes
No Is invia configured
for source zone?
Is outvia configured
for destination zone?
Is source zone different
from target zone?
Yes
Is enable-intrazone
configured?
No
No
Is the configured
viazone local or
remote?
Yes Yes
Is source zone different
from configured outvia?
Yes
Yes
No Match.
Return and continue
processing ARQ.
No
Select an IPIPGW from
the local viazone.
Send LRQ
Local Remote
Send ACF Send ARJ
Found Not found
No Is outvia configured
for destination zone?
.
Is the ARQ for an
IP-IP call?
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Chapter 7 Call Routing
Routing Using the Border Element
Figure 7-7 Single Zone for Calls
Example 7-2 Configuration of a Single Zone
dial-peer voice 1 voip
destination-pattern 91.
session target ras
incoming called-number 91.
codec transparent
gatekeeper
zone local regsite example.com 10.1.1.1
zone local vzone example.com enable-intrazone
zone remote pstnzone example.com 15.1.1.1 1719 invia vzone outvia vzone
no zone subnet vzone default enable
zone subnet vzone 10.1.1.1/32 enable
zone prefix pstnzone 91..........
no use-proxy regsite default inbound-to terminal
no use-proxy regsite default outbound-from terminal
no use-proxy vzone default inbound-to terminal
no use-proxy vzone default outbound-from terminal
no shutdown
Note Deployments may use an external NAT and Firewall device to ensure security and IP address hiding.
The border element can provide IP address hiding because it can have an outside address. However,
firewalls need to be aware of the protocol (H.323v4/5) to provide greater security by dynamically
opening holes for the calls.
SBC
WAN
IP PSTN
2
5
0
2
8
3
PSTN
ISDN
V
V
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Chapter 7 Call Routing
Routing Using the Border Element
Calls with Multiple Zones
Deployments with multiple locations are more scalable with a gatekeeper and border element per
location. With a border element per location, RSVP configurations are more manageable, and this also
helps keep the dial plan distributed. Each gatekeeper resolves calls within its location or sends calls to
other locations. The border element can be inserted into calls originated by or sent from that gatekeeper
to other locations, or for calls coming into that location. For larger deployments with multiple
gatekeepers, a directory gatekeeper can keep the dial plan simple.
The border element must be inserted when calls traverse the management domains or when connecting
to the IP PSTN from an inside network. There would then be a single trusted device for incoming and
outgoing calls through an application-aware firewall.
As illustrated in Figure 7-8, multiple locations can each have a border element in multiple respective
zones on the gatekeeper. The respective border elements are chosen for calls to and from the locations.
Media bypass can be used for intra-site calls because every call will involve the border element.
(Example 7-3 lists the configuration details.)
Figure 7-8 Multiple Zones for Calls
Example 7-3 Configuration of Multiple Zones
interface FastEthernet0/0
ip address 10.1.1.1 255.255.255.0
h323-gateway voip interface
h323-gateway voip id vzone ipaddr 10.1.1.1 1719
h323-gateway voip h323-id regsite-cube
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 10.1.1.1
Gatekeeper
SBC
WAN
IP PSTN
2
5
0
2
8
4
PSTN
ISDN
SBC SBC
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Chapter 7 Call Routing
Routing Calls to Other Networks
dial-peer voice 1000 voip
destination-pattern .T
session target ras
incoming called-number .T
codec transparent
gatekeeper
zone local regsite example.com 10.1.1.1
zone local vzone example.com enable-intrazone
zone remote pstnzone example.com 14.1.1.1 1719 invia vzone outvia vzone
zone remote vsitea example.com 11.1.1.1 1719 invia vzone outvia vzone
zone remote vsiteb example.com 12.1.1.1 1719 invia vzone outvia vzone
no zone subnet regsite default enable
zone subnet regsite 10.1.1.10/32 enable
zone subnet regsite 10.1.1.11/32 enable
no zone subnet vzone default enable
zone subnet vzone 10.1.1.1/32 enable
zone prefix vsitea 415.......
zone prefix vsiteb 650.......
gw-type-prefix 1#* default-technology
no use-proxy regsite default inbound-to terminal
no use-proxy regsite default outbound-from terminal
no use-proxy vzone default inbound-to terminal
no use-proxy vzone default outbound-from terminal
no shutdown
Note This approach can be used to migrate existing Cisco Multimedia Conference Manager (MCM) Proxy
users to the present Cisco Unified Border Element. In addition, this approach allows you to configure
call admission control as needed.
Routing Calls to Other Networks
Calls to other unified communications network can be done using their trunk interfaces. The gatekeeper
plays a key role to route calls to and from such networks. Unified communications networks can be peers
or remote gatekeepers to this video network, or they can register as gateway devices to the gatekeeper.
Call admission control by the gatekeeper is done using bandwidth configurations. The dial plan and call
routing on the gatekeeper can be done with the zone prefix commands.
Some unified communications systems need both the signaling and media for the calls to be from a single
device, or they need fixed devices to provide the call traffic. In such cases the gatekeeper can be used as
the signaling entity, and the border element can do additional call signaling and media. Appropriate
devices must be chosen to support the scale according to device performance and capabilities.
Routing Calls Through Untrusted Networks
When enterprises need to communicate across untrusted networks, they use firewalls as security devices
to prevent unauthorized access to internal networks. Firewalls can do protocol inspection of VoIP
protocols to provide a similar level of security for calls. Firewalls can inspect VoIP protocols, H.323, or
SIP signaling messages and allow the appropriate traffic for the media to go between the two entities.
The border element adds value here by being the one trusted device that the firewall can inspect for call
signaling traffic to and from the internal network. This not only reduces the firewall configuration
complexity but also allows the border element to do the topology hiding for the internal network.
Calls then can be routed through external public gatekeepers or directly to other enterprises based on the
dial plan configuration. External untrusted endpoints must register with an enterprise gatekeeper so that
the registration and dial plan can be managed by the enterprise. Gatekeepers then can restrict
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Chapter 7 Call Routing
Routing Calls to Other Networks
unauthorized registrations; the firewall can provide protocol inspection and security against
unauthorized calls; and the border element can provide number manipulation, load balancing, call
admission control, and RSVP if configured.
C H A P T E R
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8
Cisco Video Infrastructure Components
Last revised on: October 30, 2009
This chapter describes the Cisco Unified Videoconferencing infrastructure components and related
network design considerations. This chapter focuses on traditional (room-based) IP videoconferencing.
Therefore, it is specifically intended for those environments that have yet to migrate to an end-to-end
Cisco Unified Communications Manager Video Telephony model or Cisco Unified MeetingPlace Video
Integration.
For information on integrating the Cisco Unified Videoconferencing infrastructure components with
Cisco Unified Communications Manager, refer to the IP Video Telephony section in the Cisco Unified
Communications Solution Reference Network Design (SRND), available at
http://www.cisco.com/go/ucsrnd
Note MCU and gateway features may change with new software releases, and those changes might not be
represented in this document. Refer to the latest product documentation and release notes for details.
What's New in This Chapter
Table 8-1 lists the topics that are new in this chapter or that have changed significantly from previous
releases of this document.
Table 8-1 New or Changed Information Since the Previous Release of This Document
New or Revised Topic Described in:
Collaboration Collaboration for Desktop Sharing, page 8-21
Connecting to external networks Connecting the Video Network with External
Untrusted Networks, page 8-27
Integration with other collaboration applications Integration and Interoperability, page 8-25
Multipoint conferencing Multipoint Conferencing, page 8-2
Recording conferences Conference Recording, page 8-24
Scheduling video conferences with Cisco WebEx Video Scheduling with Cisco WebEx, page 8-8
Streaming conference video Conference Streaming, page 8-23
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Chapter 8 Cisco Video Infrastructure Components
Design and Deployment Overview
Design and Deployment Overview
Video infrastructure design is a very important element in an H.323, Session Initiation Protocol (SIP),
or Skinny Client Control Protocol (SCCP) videoconferencing network. In the H.320 circuit-switched
network, Multipoint Control Units (MCUs) and H.320 endpoints are connected directly to the PSTN
network. In the past, an MCU could have multiple PRI connections into the switched network. The
switched network supplied a dedicated transport with guaranteed bandwidth and predictable delay. Now
that video is being moved onto IP networks that share bandwidth with data, placement of video
infrastructure components becomes very important. Installing a central MCU and/or gateway in an IP
environment does not always work. Bandwidth in an IP network is not dedicated to each video device
on the network, therefore it is important to design the network accordingly.
The design of systems with Cisco Unified Videoconferencing components is best illustrated by
reviewing design scenarios step-by-step. This method provides an understanding of how best to deploy
Cisco Unified Videoconferencing for your particular environment. The scenarios in this chapter are
designed to simplify the end-user experience and conserve bandwidth. Therefore, no matter what stage
your videoconferencing network is in today, you can benefit from deploying one or more of the
videoconferencing scenarios described in this chapter.
Multipoint Conferencing
Whenever three or more parties join a videoconference, a Multipoint Control Unit (MCU) is required.
The MCU can mix all video streams together and transmit a composite image of all participants back to
the originating sources. This composite view (called continuous presence) is necessary for all
participants to see each other simultaneously. The continuous presence view can display from 2 to 32
windows (participants) in a variety of different layouts. Each layout offers the ability to make one of the
windows voice-activated, which is useful if there are more participants in the conference than there are
windows to display them all in the composite view. More than 32 participants can be in a single
videoconference, and the MCU will allow the last active sites to be displayed. Moreover, as participants
join or leave the conference, the Cisco Unified Videoconferencing MCU can automatically adjust the
continuous presence view by dynamically changing the layout.
Continuous presence can have multiple participants using different video and audio codecs. In this
situation, the MCU must reconstitute all the video streams into a single image while at the same time
preserving the video and audio codec of each participant. In addition, the MCU can save network
bandwidth by avoiding the need for the sources to transmit large video streams to all sources. Although
MCUs can be implemented as either hardware or software, hardware-based MCUs guarantee superior
video quality by performing advanced transcoding, transrating, and composition features.
The Cisco Unified Videoconferencing solution consists of the following main components:
Cisco IOS Gatekeeper
This component provides address resolution and H.323 audio and video setup and tear down.
Cisco Unified Videoconferencing MCU
This component performs call signaling and multipoint processing of all audio. In addition, the
MCU hosts the web interface and controls one or more Enhanced Media Processors (EMPs).
EMP
The EMP is a dedicated multipoint processor for video. The EMP cannot function without a
controlling MCU.
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Chapter 8 Cisco Video Infrastructure Components
Multipoint Conferencing
H.320 Gateway
This component bridges ISDN video endpoints into the IP H.323 infrastructure. There are several
ways to use the ISDN gateways in a video solution.
Desktop Streaming Server
The Desktop streaming server streams the conference to user desktop PCs or laptops. The Desktop
server provides the collaboration features for video conferences by allowing users to share desktop
or laptop screens and by providing enhanced roster and conference moderation functionality. It also
enables conferences to be streamed using Real Time Streaming Protocol (RTSP), to be viewed
through a media player or browser plug-in as a webcast.
Desktop Recording Server
The Recording server can provide recording of the videoconference and desktop sharing. It can also
be a repository for the recorded meetings to be viewed later.
Management and Integration Server
The Management server can be the central entity for configuration and management of the various
video devices and the videoconferencing elements. Thus, it can be used to configure, manage, and
monitor devices such as MCUs and gatekeepers, among other. The server can also perform
conference scheduling, be a gatekeeper, and provide cascading logic for video conferences between
different locations to optimize WAN utilization if possible.
The Integration servers allow the enterprise calendaring systems to assign and reserve resources
such as MCU ports for the conference, thus ensuring that sufficient resources are available during
the conference.
The recommended multipoint hardware platforms are the Cisco Unified Videoconferencing 5000 Series,
3545 and 3515 MCUs. With the Cisco Unified Videoconferencing 3545 and 3515 MCUs, video
processing is done by the EMP module, and the MCU performs audio processing and conference control.
The Cisco Unified Videoconferencing 3515 MCU is designed to be a self-contained unit for smaller
deployments. Therefore, it has an integrated EMP along with an integrated MCU inside a fixed chassis
that cannot be upgraded in the field. By contrast, the Cisco Unified Videoconferencing 5200 or 3545
System is modular and is designed for maximum deployment flexibility. Therefore, MCUs, EMPs, or
H.320 gateway modules can be inserted into the Cisco Unified Videoconferencing 5200 or 3545 chassis
in order to process audio and video, and this system can be upgraded in the field.
The Cisco Unified Videoconferencing 5000 Series and 3500 Series MCUs support H.323 (standard
definition and high definition), SCCP, and SIP endpoint interoperability. The encoder-per-port hardware
architecture allows any supported connection speed from 64 kbps up to 2 Mbps, with any supported
codec and at any standard-definition resolution. Supported protocols include H.261, H.263, and H.264
video codecs, as well as G.711, G.722, G.722.1, G.723.1, G.728, and G.729A audio codecs.
High-definition video calls provide better video to users. Using traditional or desktop-based
high-definition endpoints can enhance the conference experience. High-definition video also requires
that the network must be able to process the greater volume of traffic for the calls and must provide
Quality of Service (QoS), traffic classification, and call admission control where network resources are
not sufficient. For additional information on multipoint conferencing, continuous presence, and voice
activation, refer to the IP Video Telephony section in the Cisco Unified Communications Solution
Reference Network Design (SRND), available at
http://www.cisco.com/go/ucsrnd
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Chapter 8 Cisco Video Infrastructure Components
Initiating a Call
Initiating a Call
To initiate a multipoint call using a Cisco Unified Videoconferencing MCU, the endpoint dials the
appropriate service prefix followed by a conference ID. (The conference ID can be up to 9 digits long.)
If a service on an MCU is 40880 for a 384-kbps call, the user might dial 4088011223, the call would be
routed to the MCU using the 40880 service prefix, and the MCU would initiate an ad hoc conference
with an ID of 4088011223. Users can also initiate a call and invite the other participants by dialing the
conference ID, the invite string **, and the E.164 address of the another participant. Figure 8-1 shows
the dial sequence of an MCU call with Video 1 initiating an MCU call to 40880123 and inviting Video
2 and Video 3.
Figure 8-1 Example Dial Sequence for Initiating a Call (MCU Invite Call)
In addition, it is possible for an H.323 endpoint to dial the MCU by IP address. The MCU will answer
the call with an audio IVR and an on-screen menu.
Dial string
40880123**4085456780**4085456089
Video 1
E.164 = 4085455565
Video 2
E.164 = 4085456780
Video 3
E.164 = 4085456089
Conference
40880123
QoS Enabled
IP LAN
8
1
0
4
4
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Chapter 8 Cisco Video Infrastructure Components
Cascading MCUs
Cascading MCUs
Cascading MCUs allows larger conferences to be created by combining resources from multiple MCU
blades or units. Cascading is also used in distributed MCU environments to save bandwidth on low-speed
WAN links (see Distributed MCUs, page 8-5). Cisco Unified Videoconferencing MCUs support
cascading. An administrator can cascade MCUs by inviting conference calls on different MCUs to join
in a single combined call. In Figure 8-2, a conference was started on each of the four MCUs. To cascade
the MCUs in this example, an administrator accessed the web interface on MCU 1 and invited
conferences 4088112 on MCU 2, 4088632 on MCU 3, and 4088552 on MCU 4.
Figure 8-2 Cascaded MCU Conference
Note There is no physical connection made between the cascaded devices. Cascading occurs over the LAN or
WAN, allowing MCUs to be distributed across a network. An MCU can invite H.323 endpoints, H.320
endpoints through a gateway, or other MCUs through the web interface on the MCU.
Distributed MCUs
With the ability to cascade multipoint conferences, administrators can build an H.323 video network
with distributed MCU services. A distributed MCU architecture saves WAN bandwidth when a
conference includes multiple participants on two or more campuses connected by a WAN. By
distributing MCUs across the network, it is possible to have multipoint conferences across WAN links
without limiting the number of users at remote sites.
Centrally located MCU services require all conference participants to place a call across the WAN to the
MCU, while distributed MCUs allow users to call to their local MCU and join the other MCUs into a
cascaded conference across the WAN. Figure 8-3 illustrates centralized MCU services, with three users
from Campus B joining a conference hosted at Campus A. In this model, all three calls must traverse the
WAN link.
MCU 4
Conference
40880552
MCU 2
MCU 3
MCU 1
Conference
40880112
Conference
4088632
Conference
4088222
QoS Enabled
IP LAN
8
1
0
4
5
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Chapter 8 Cisco Video Infrastructure Components
Dynamic Cascading of MCUs
Figure 8-3 Centralized MCU Services, with All Calls Traversing the WAN
Figure 8-4 illustrates a distributed MCU model with two video terminals at Campus A calling into a local
MCU, and three video terminals at Campus B calling into a local MCU. In this model, there is just a
single call cascading the two conferences across the WAN. In most distributed networks, a Cisco Unified
Videoconferencing 5000 or 3545 System will be located at the headquarters site and a Cisco Unified
Videoconferencing 3515 MCU will be located at each large remote site.
Figure 8-4 Distributed MCU Services, with a Single Call Traversing the WAN
Dynamic Cascading of MCUs
Dynamic cascading improves upon normal cascading by allowing conferences to be distributed
automatically to local MCUs, with no administrator or participant intervention. By using single number
reach, dynamic cascading hides the complexity from participants and automatically combines
distributed MCUs from multiple locations. Dynamic cascading works through automatic scheduling and
intelligent assignment of distributed MCU resources. Participants join a single combined call that
reduces the network bandwidth consumption to only a single video call across the WAN.
As the conference grows in size, dynamic cascading will intelligently expand the conference as long as
sufficient network resources are available. The Cisco Unified Videoconferencing 5000 Series or
3500 Series MCUs support dynamic cascading. In Figure 8-5, a conference was automatically started on
each of the four MCUs. The same number (4088222) was dialed by all eight participants, but dynamic
cascading automatically assigned the local terminals to the respective local MCUs. The option is
available for the MCUs to dial out directly to the terminals, thereby simplifying the user experience.
Three H.323 Video
calls across WAN
QoS Enabled
IP WAN
Campus
A
Campus
B
8
1
0
4
6
One H.323 call across WAN
QoS Enabled
IP WAN
Campus
Campus
8
1
0
4
7
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Chapter 8 Cisco Video Infrastructure Components
Scheduled Conferencing
Figure 8-5 Dynamically Cascaded MCU Conference
Note Dynamic cascading requires a Cisco IOS Gatekeeper and a videoconferencing scheduler using either
Cisco Unified MeetingPlace Video Integration or Cisco Unified Videoconferencing Manager.
Note When deployments have a combination of Cisco Unified Videoconferencing 5000 Series or 3500 Series
MCUs, cascading between the two series types of MCUs must be done manually per conference.
However, dynamic cascading is supported between MCUs of the same series type.
Scheduled Conferencing
Scheduled videoconferencing provides the ability to reserve resources in advance, as well as to simplify
the videoconferencing experience for end users. Administrators can grant users the ability to schedule
conferences through email integration or a web interface. The scheduling interface hides all
videoconferencing complexity from the end user.
There are three applications used to provide scheduling videoconferencing in a Cisco Unified
Videoconferencing environment:
Cisco Unified MeetingPlace with Video Integration
Cisco Unified Videoconferencing Manager
Cisco WebEx with Cisco Unified Videoconferencing integration
Video Scheduling with Cisco Unified MeetingPlace
Cisco Unified MeetingPlace is the recommended method for scheduling video conferences. The
participants schedule video resources through email (Microsoft Outlook or Lotus Notes) or a
simple-to-use web interface. Web conferencing can easily be added to provide integrated voice, video,
and web conferencing solutions that fit any enterprise.
MCU 1
Conference
4088222
2
5
0
4
1
6
MCU 3
Conference
4088222
Qos Enabled
IP WAN
MCU 2
Conference
4088222
MCU 4
Conference
4088222
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Chapter 8 Cisco Video Infrastructure Components
Scheduled Conferencing
For additional information on Cisco Unified MeetingPlace Video Integration, including detailed video
call flows, refer to the Cisco Unified MeetingPlace section of the Cisco Unified Communications
Solution Reference Network Design (SRND), available at
http://www.cisco.com/go/ucsrnd
Video Scheduling with Cisco WebEx
Enterprises that use Cisco WebEx for collaboration can use the WebEx scheduling features for their
conferences. WebEx scheduling is done through the cloud services provided by WebEx. WebEx
scheduling provides integration for various email systems through plug-ins. The plug-ins enable users
to schedule conferences using the enterprise calendaring systems and to choose users and resources that
then get updated in the WebEx cloud through the users WebEx accounts.
WebEx integrations with the MCU and with Cisco Unified Videoconferencing Desktop allows the
enterprise to provide the needed QoS for the scheduled calls and for integration with user endpoints. For
additional details on the integration, see the section on Integration and Interoperability, page 8-25.
Video Scheduling with Cisco Unified Videoconferencing Manager
For environments without Cisco Unified MeetingPlace, scheduling is accomplished with Cisco Unified
Videoconferencing Manager, which is a standalone videoconferencing option only. It enables users to
schedule voice and video conferences. Cisco Unified Videoconferencing Manager can be used with the
Outlook integration option or with web-based scheduling. Moreover, synchronization with LDAP can
minimize user setup and maintenance as well as providing authentication. In addition, LDAP can be used
to integrate video terminals into the LDAP directory and associate Class of Service policies to users.
Class of Service (CoS) is accomplished by restricting users to specific meeting types. For example,
administrators can grant users the ability to schedule meetings with only standard-rate video service
(384 kbps video streams). Likewise, administrators can grant another set of users the ability to schedule
meetings with high-definition video service (up to 2 Mbps).
Note Cisco Unified Videoconferencing Manager can provide call detail records (CDRs) for all
videoconferencing calls.
Figure 8-6 illustrates Cisco Unified Videoconferencing Manager together with a Cisco IOS Gatekeeper.
Using Microsoft Outlook, a participant in this example has scheduled the video terminals into a
conference. When the meeting starts, Cisco Unified Videoconferencing Manager automatically out-dials
all terminals into the call. All of this occurs without any user intervention aside from using Outlook to
schedule the video conference. Optionally, Cisco Unified Videoconferencing Manager can also
dynamically cascade all terminals. Thus, bandwidth consumption on the network is minimized to a
single call.
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Chapter 8 Cisco Video Infrastructure Components
Scheduled Conferencing
Figure 8-6 Cisco Unified Videoconferencing Manager
Note Video media streams do not flow through Cisco Unified Videoconferencing Manager. Video media
terminates on the MCUs.
Virtual MCU
Cisco Unified Videoconferencing Manager has the capability to allow all MCU resources to appear as a
single MCU. This virtual MCU concept provides a single conference ID across multiple MCUs.
Therefore, this reduces the complexity of scheduling because participants can dial a single number and
automatically be associated to the closest MCU. Likewise, the system can dial out to the participants.
The virtual MCU provides the logic for dynamic cascading.
In Figure 8-7, the same conference call illustrated in Figure 8-6 is now depicted as a purely logical
entity. The four distributed MCUs become a single virtual MCU when using Cisco Unified
Videoconferencing Manager for scheduling the video conference. Participants can dial the number
4088222 located on the meeting invitation, and Cisco Unified Videoconferencing Manager will assign
video terminals to local MCU resources. The virtual MCU has intelligent topology awareness of local
resources, so it can avoid unnecessary bandwidth congestion over the WAN.
MCU 1
Conference
4088222
2
5
0
4
1
7
MCU 3
Conference
4088222
Qos Enabled
IP WAN
MCU 2
Conference
4088222
MCU 4
Conference
4088222
Gatekeeper 1
Cisco Unified
Videoconferencing
Manager
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Chapter 8 Cisco Video Infrastructure Components
Scheduled Conferencing
Figure 8-7 Virtual MCU Created by Cisco Unified Videoconferencing Manager
Note To enable dynamic cascading, Video terminals and MCUs must be associated to a location in the
Resource Manager component of Cisco Unified Videoconferencing Manager.
Cisco Unified Videoconferencing Manager Components
Cisco Unified Videoconferencing Manager contains two major software components:
Resource Manager A scheduling component
Network Manager A network management component
Thus, Cisco Unified Videoconferencing Manager combines a scheduling application with a network
management application.
Resource Manager Component of Cisco Unified Videoconferencing Manager
The Resource Manager component of Cisco Unified Videoconferencing Manager controls all aspects of
videoconferencing and interfaces with MCUs and the gatekeeper. Resource Manager is the main control
point for all incoming and outgoing video calls, and it makes all decisions regarding resource utilization
and cascading of resources. In addition to basic MCU cascading to increase conference size, Resource
Manager can intelligently select which MCU to use based on internally defined locations for each
participant. Multisite video meetings can result in cascading MCUs local to user groups, thus creating
one link across a WAN between cascaded MCUs.
The Resource Manager component of Cisco Unified Videoconferencing Manager has the following
characteristics:
Resource Manager resides on a separate Windows-based server and cannot reside on the same server
with other Cisco Unified Communications components.
Resource Manager sits between the MCUs and all other components.
2
5
0
4
1
8
Gatekeeper 1
Cisco Unified
Videoconferencing
Manager
Virtual MCU
4088222
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Chapter 8 Cisco Video Infrastructure Components
Scheduled Conferencing
Resource Manager contains a special integrated gatekeeper to which only the MCUs register. The
MCUs must register to Resource Manager.
All routing decisions for MCU selection and cascading are made by Resource Manager.
Resource Manager terminates and authenticates all inbound calls, and only media is sent to the
MCUs.
Resource Manager originates and maintains all outbound calls, and only media is sent to the MCUs.
Resource Manager requires the use of an external gatekeeper for outbound calls.
Resource Manager does not register to the gatekeeper it uses for outbound calls.
Resource Manager does not support a direct SIP trunk connection from Cisco Unified
Communications Manager.
Resource Manager does not support a direct H.323 gatekeeper-controlled trunk connection from
Cisco Unified Communications Manager.
All scheduled video meetings are replicated in Resource Manager, with appropriate resources
reserved.
Participant information and real-time status are relayed from the MCUs to Resource Manager.
Other than media stream termination, only Resource Manager communicates with the MCUs.
Video Terminals can be defined in Resource Manager and reserved within Cisco Unified
Videoconferencing Manager when users create a meeting.
Any endpoint (H.323, SCCP, or SIP) can be defined in Resource Manager as a Video Terminal.
MCU selection can be impacted by assigning locations to MCUs and Video Terminals defined in
Resource Manager.
Note The MCUs register to the Resource Manager internal gatekeeper as an MCU instead of as a gateway. The
parameter is located on the MCU under Advanced H.323 Settings.
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Chapter 8 Cisco Video Infrastructure Components
Scheduled Conferencing
Video Endpoint and MCU Registration
When using Cisco Unified Videoconferencing Manager, all H.323 video endpoints should be configured
to register to the Cisco IOS gatekeeper. Figure 8-8 shows all H.323 video endpoints configured with the
Cisco IOS gatekeeper as their designated gatekeeper.
Figure 8-8 Video Endpoint H.323 Registration to a Cisco IOS Gatekeeper
You must register Cisco Unified Videoconferencing MCUs and Cisco Unified Videoconferencing
Gateways with the Cisco Unified Videoconferencing Manager internal gatekeeper, which is included
with the product. This preserves the virtual MCU features and conference setup, and it allows for
conference control via a web interface during a meeting.
Figure 8-9 shows all Cisco Unified Videoconferencing MCUs configured with the Resource Manager
internal gatekeeper as their designated H.323 gatekeeper.
2
5
0
4
1
9
Gatekeeper 1
Cisco Unified
Videoconferencing
Manager
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Scheduled Conferencing
Figure 8-9 MCU H.323 Registration to Cisco Unified Videoconferencing Manager
For integration of video endpoints or Cisco Unified Videoconferencing MCUs with Cisco Unified
Communications Manager, refer to the IP Video Telephony section in the Cisco Unified Communications
Solution Reference Network Design (SRND), available at
http://www.cisco.com/go/ucsrnd
Gatekeeper Integration with Resource Manager
The Resource Manager internal gatekeeper and the Cisco IOS Gatekeeper are configured as H.323
gatekeeper neighbors.
Example 8-1 illustrates the minimal configuration necessary for the Cisco IOS gatekeeper to integrate
with the Cisco Unified Videoconferencing Manager. The Resource Manager internal gatekeeper is a
remote zone from the perspective of the Cisco IOS gatekeeper.
Example 8-1 Cisco IOS Gatekeeper Minimal Configuration with Cisco Unified Videoconferencing
Manager
gatekeeper
zone local HQ cisco.com 10.1.1.1
zone remote CUVM cisco.com 10.2.2.1 1719
zone prefix CUVM 83* (This entry matches service prefixes configured
on the MCUs)
gw-type-prefix 1#* default-technology
lrq forward-queries add-hop-count
no use-proxy HQ remote-zone CUVM inbound-to terminal
no use-proxy HQ remote-zone CUVM outbound-from terminal
no use-proxy HQ default inbound-to terminal
no use-proxy HQ default outbound-from terminal
no shutdown
When integrating Resource Manager into an infrastructure gatekeeper environment with Cisco Unified
Communications Manager (Unified CM), the deployment options discussed in this section will bypass
Unified CM when one H.323 endpoint calls another H.323 endpoint. Having the endpoints controlled by
Unified CM is a deployment option not covered here due to the focus on environments that have yet to
2
5
0
4
2
0
Gatekeeper 1
Cisco Unified
Videoconferencing
Manager
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Chapter 8 Cisco Video Infrastructure Components
Scheduled Conferencing
deploy Cisco Unified CM. For information on integrating Cisco Unified CM with video endpoints, refer
to the IP Video Telephony section in the Cisco Unified Communications Solution Reference Network
Design (SRND), available at
http://www.cisco.com/go/ucsrnd
Resource Manager Redundancy Considerations
Resource Manager is limited to server component redundancy, and it does not have any software-level
redundancy. The server on which Video Administration is deployed should contain redundant
components to minimize the risk of downtime. Cisco Unified Videoconferencing Manager is deployed
on an MCS-7825 server regardless of the total number of endpoints, MCUs, or ISDN gateways under
control.
Redundancy of the Cisco IOS Gatekeeper via Hot Standby Router Protocol (HSRP) provides for
outbound calling redundancy from Resource Manager. If the primary gatekeeper fails, call resolution
requests are automatically sent to the backup gatekeeper. Resource Manager does not need to know about
the backup gatekeeper or primary gatekeeper state to accomplish this failover.
Gatekeeper Clustering and Alternate Gatekeeper
With the gatekeeper clustering and alternate gatekeeper methods, an issue arises with outbound calls
from Resource Manager. Outbound calls from Resource Manager are sent to the Cisco IOS Gatekeeper
without any registration to that gatekeeper. This prevents Resource Manager from being aware of the
gatekeeper state and the existence of an alternate gatekeeper.
Network Manager Component of Cisco Unified Videoconferencing Manager
Cisco Unified Videoconferencing Manager can provide network management of MCUs and video
endpoints. The Network Manager component of Cisco Unified Videoconferencing Manager queries the
Cisco IOS Gatekeeper to discover all video endpoints on the network. This allows organizations to have
a global view of MCUs, gatekeepers, and video endpoints. This video infrastructure can change
dynamically yet be discovered automatically. Network Manager can also provide software upgrades to
MCUs and many third-party endpoints.
In addition, the Network Manager component can provide real-time status of MCU health, alarms,
events, active calls, and active conferences. Moreover, recipients can be alerted to events through email,
and SNMP traps can be forwarded to other management systems.
Cisco Unified Videoconferencing Manager Deployment Considerations
There are three general types of deployments for Cisco Unified Videoconferencing Manager:
Scheduling mode
This mode allows scheduling integration with Microsoft Outlook and simplifies all aspects of video
conferencing setup. This mode can be further categorized as with or without dynamic cascading.
Network Management mode
The network management features of Cisco Unified Videoconferencing Manager can be used solely
for endpoint and MCU management without any participation in scheduling or dynamic cascading.
Scheduling mode with Network Management mode
This method utilizes all of the components of the Cisco Unified Videoconferencing Manager.
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Chapter 8 Cisco Video Infrastructure Components
Video Gateways
Video Gateways
The Cisco Unified Videoconferencing 3545 Dual PRI, 3527 Single PRI, and 3522 4-Port BRI Gateways
give enterprises the ability to connect ISDN-based H.320 systems with IP-based H.323 videoconference
endpoints. These gateways provide translation services between H.320 and H.323 networks to convert
multimedia information between circuit-switched ISDN and IP networks. The gateway also supports
G.711, G.722, G.722.1, G.723.1, and G.728 audio codecs. In addition, voice transcoding between IP and
the Public Switched Telephone Network (PSTN) is supported using G.711, G.723, or G.728. These
systems enable users to videoconference with others users via the LAN or the PSTN, regardless of
location. The Cisco 3500 Series Gateways also provide in-band DTMF conversion to out-of-band
DTMF.
The three video gateway models provide the following features:
Cisco Unified Videoconferencing 3522 BRI Gateway
This gateway can be configured with two or four BRI ports. When the gateway is equipped with BRI
ports, it can support calls up to 384 kbps on aggregated channels.
Cisco Unified Videoconferencing 3527 PRI Gateway
This gateway is a self-contained system that supports a high volume of calls over a single high-speed
ISDN PRI connection, allowing dynamic allocation of its 23 B channels.
Cisco Unified Videoconferencing 3545 Gateway Modules
These gateway modules are either a two-port PRI T1/E1 module that supports a high volume of calls
over multiple high-speed ISDN PRI connections, or a four-port serial module for IP connectivity to
older ISDN H.320 videoconferencing endpoints.
Table 8-2, Table 8-3, and Table 8-4 list the maximum number of calls supported per platform.
Table 8-2 Cisco Unified Videoconferencing 3522 BRI Gateway Call Handling Capacity
Number of Calls Capacity
1 384 kbps or 412 kbps
2 256 kbps
4 128 kbps
8 64 kbps
Table 8-3 Cisco Unified Videoconferencing 3527 PRI Gateway Call Handling Capacity
Call Type
Maximum Number of Calls Using
One E1 PRI Line
Maximum Number of Calls Using
One T1 PRI Line
Voice (64 kbps) 30 23
2B video (128 kbps) 15 11
6B video (384 kbps) 5 3
12B video (768 kbps) 2 1
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Chapter 8 Cisco Video Infrastructure Components
Outbound Dialing Service Prefixes
Outbound Dialing Service Prefixes
Video gateways must be configured with service prefixes to define the speed of outgoing calls and
calling routing to the video gateway. In telephony systems, dialing 9 to access an outside line is very
common. In order to keep dialing strings consistent with existing voice dial plans, Cisco recommends
using 9# for video gateway service prefixes. Using the # in the service prefix ensures that ISDN users
do not access the IVR and hairpin the call back out the ISDN network. The # is used as a delimiter by
the gateway and prevents hair pinning from the ISDN network.
From the users perspective, they will have to dial the service prefix, which in this case is equivalent to
an access code, followed by the ISDN number of the H.320 video unit. For this configuration, a local
(intra-zone) gateway is used whenever one is present. In zones that do not contain a gateway, the
administrator should assign a gateway in another zone as the primary gateway for the local zone.
Configure location request (LRQ) forwarding or a static hopoff statement to route all calls to a zone with
a gateway for PSTN access.
Network Load Balancing
The Cisco Unified Videoconferencing Gateway supports the network load balancing (LAN to PSTN)
feature. This feature allows users to build a pool of gateways for PSTN access. Network load balancing
creates a larger number of access lines serviced by a single set of service prefixes.
Gatekeepers can perform load balancing on the network using feedback from the gateway in the form of
Resource Availability Indication (RAI) messages that inform the gatekeeper of gateway resource
availability. If the gateway is unavailable, the gatekeeper performs line hunting operations to route the
call to an alternate gateway. When you set the gateway for RAI and Resource Availability Confirmation
(RAC), it sends periodic RAI messages that inform the gatekeeper of the current resource availability in
the gateway. The gatekeeper responds with Resource Available Confirmation (RAC) messages to
acknowledge receipt of the RAI messages.
To implement network load balancing, you configure multiple gateways with identical service prefixes
and register them with the same gatekeeper. Outbound PSTN calls are sent to the gateways based on
resource availability, using RAI and RAC. In the gateway configuration, you set utilization parameters
based on gateway resource percentages.
The main gateway configuration parameters for line hunting are:
Utilization (percent load) for sending RAI ON message
Utilization (percent load) for sending RAI OFF message
Table 8-4 Cisco Unified Videoconferencing 3545 PRI Gateway Call Handling Capacity
Call Type
Maximum
Number of Calls
Using One E1 PRI
Line
Maximum
Number of Calls
Using One T1 PRI
Line
Maximum
Number of Calls
Using Two E1 PRI
Lines
Maximum
Number of Calls
Using Two T1 PRI
Lines
Voice (64 kbps) 30 23 60 46
2B video (128 kbps) 15 11 30 23
6B video (384 kbps) 5 3 10 7
12B video (768 kbps) 2 1 5 3
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Cisco Unified Border Element
The RAI ON message tells the gatekeeper that resources are running low on the gateway that sent the
message, and the gatekeeper should not forward any more calls to that gateway. (The default for sending
a RAI ON is 80% load.) The RAI OFF message tells the gatekeeper that there are enough available
resources on the gateway, and calls can be forwarded to the gateway again. (The default for sending a
RAI off is 60% load.) Periodic RAI messages are sent from the gateway to the gatekeeper when one of
the above thresholds is not achieved in a specified period of time (The default period for these messages
is 30 seconds.)
Cisco Unified Border Element
The Cisco Unified Border Element is a Cisco IOS software component that incorporates the gatekeeper
and border element functions for an H.323 video network. The Cisco IOS gatekeeper provides endpoint
registration and call resolution for large H.323 video networks. The border element provides topology
hiding, network isolation, call admission control and Quality of Service (QoS). The border element is
increasingly used for dynamic call admission control with RSVP devices or on behalf of devices that do
not support RSVP. Deployments with a border element simplify Network Address Translation (NAT) and
firewall integration.
Gatekeeper
The Cisco gatekeeper performs all call routing and address registration (RAS) for all H.323 video
components. The gatekeeper is one of the most important components in an H.323 network because it is
the central management device for the H.323 video network and it performs functions required for a
successful H.323 video deployment. Some of the most commonly used functions of the Cisco IOS
gatekeeper include:
H.323 component registration and call routing
The gatekeeper registers the IP address, E.164 address, H.323-ID, device type, and signaling ports
for all the video infrastructure components. This registration allows the gatekeeper to provide call
routing for all devices that are registered with the it.
Bandwidth management
Managing video bandwidth on IP networks is an essential feature of any gatekeeper. By setting the
following bandwidth parameters, you can configure the Cisco gatekeeper to manage the bandwidth
in a zone, between zones, or per call.
Inter-zone: Total bandwidth allowed from a local or default zone to and from all other zones.
Remote: Total bandwidth allowed from all local zones to and from all remote zones.
Session: Bandwidth allowed per session in a zone.
Total: Total bandwidth allowed in a zone.
Resource Availability Indicator (RAI): Endpoints supporting RAI update the gatekeeper of
availability or unavailability.
Circuit-id: Calls with this circuit-id call identifier are limited by the gatekeeper to max-calls per
endpoint.
Max-calls: Maximum number of calls that the gatekeeper will permit per endpoint.
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Cisco Unified Border Element
Authentication, authorization, and accounting (AAA) support
The Cisco gatekeeper works in conjunction with Remote Access Dial-In User Service (RADIUS)
and Terminal Access Controller Access Control System (TACACS) servers to provide
authentication of devices and accounting via call detail recording (CDR).
Via-zone
The gatekeeper can include the border element in the call flow by using the via-zone with invia and
outvia configurations to include the border element for incoming calls or outgoing calls.
The Cisco gatekeeper also supports features such as the following, which enable users to build reliable
and scalable H.323 networks:
Hot Standby Router Protocol (HSRP) enables administrators to build a standby gatekeeper that
becomes active if the primary gatekeeper fails.
Gatekeeper Update Protocol (GUP) enables multiple gatekeeper entities to behave as a single
gatekeeper cluster, providing greater resilience for endpoint registrations and calls. (Limits on
scalability must be taken into consideration.)
Directory Gatekeeper, or Location Request (LRQ) forwarding, enables administrators to build large
multi-tier networks, minimizing the configuration required in the lower-tier gatekeepers. When a
call is made in a lower-tier zone and a match is not found, the call is automatically forwarded up to
the directory gatekeeper for resolution. (For more information on directory gatekeepers, refer to the
Call Routing chapter.) Figure 8-10 on page 8-20 illustrates a network configured with two regional
directory gatekeepers, one at Site E and another at Site F.
Redundancy
A video network needs redundancy for the endpoints to register and for call resolution. Failure in call
control devices can impact the ability to do videoconferencing. Any of the following methods can
provide redundancy:
Hot Standby Router Protocol (HSRP)
HSRP can be used to provide redundancy at the IP layer for the gatekeeper. Thus, endpoints can use
one IP address for the gatekeeper address. Failures of devices providing hot standby for the
gatekeeper IP address do not impact the endpoints drastically. The blackout period after the failure
of the primary device and until the endpoints are fully registered and functional with the standby
device, depends on the device time-outs and registration retries. This method is recommended for
the directory gatekeeper because its main function is call resolution and it might not have any
endpoints registered. (For more information on directory gatekeepers, see Routing Inter-Zone Calls
Using a Directory Gatekeeper, page 7-9.)
Alternate gatekeeper
One of the simplest methods of providing redundancy is to configure the endpoints with a list of
gatekeepers in order of preference, so that if the first gatekeeper in the list fails, the devices register
with the next gatekeeper in the list. However, not all endpoints have this support. Cisco Unified
Communications Manager, Cisco Unified Border Element, and Cisco Gateways support alternate
gatekeeper functionality.
Gatekeeper Update Protocol (Cisco GUP)
The Cisco Gatekeeper Update Protocol (GUP) enables gatekeepers in the cluster to update
information on registrations and calls in the cluster. Alternate gatekeeper support is needed in the
endpoints. When an endpoint registers or unregisters with its primary gatekeeper, that gatekeeper
updates the remaining gatekeepers in the cluster about the change. Similar information is shared
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Chapter 8 Cisco Video Infrastructure Components
Cisco Unified Border Element
when the gatekeeper resolves a call. If any gatekeeper in the cluster fails, the endpoint registers to
the alternate gatekeeper that the primary gatekeeper had informed when the endpoint initially
registered. With GUP, if a gatekeeper failure occurs, active calls are not disconnected. GUP is
recommended where the number of endpoints is not very large. However, most video endpoints do
not support alternate gatekeeper, therefore they cannot support the use of GUP.
Scalability
Call routing scalability can be achieved by using the hierarchical model. Common regions or areas are
grouped together to be serviced by access gatekeepers which then point to a directory gatekeeper. The
directory gatekeeper may not have any endpoints registered to it and may have a role of routing calls
between access gatekeepers or to hop-off zones. The access gatekeeper may have endpoints registered
to it or, in larger deployments, may have an additional level of gatekeepers to which it can route calls.
HSRP
Hot Standby Router Protocol (HSRP) enables a set of routers with the Cisco Unified Border Element to
work together as a single virtual gatekeeper or border element. You can implement this feature by
creating a phantom router that has its own IP and MAC addresses.
Based on the priority given by the network administrator, one of the HSRP gatekeepers in each group is
selected to be active and the other to be standby. The gatekeeper with the highest priority serves as the
active gatekeeper. The active gatekeeper does the work for the HSRP phantom. If an end node sends a
packet to the phantom's MAC address, the active gatekeeper receives that packet and processes it. If an
end node sends an Address Resolution Protocol (ARP) request for the phantom's IP address, the active
gatekeeper replies with the phantom's MAC address.
The HSRP gatekeepers (both active and standby) watch for hello packets to monitor the status of each
other. The gatekeeper group learns the hello and hold timers, as well as the standby address to be shared,
from the active gatekeeper. If the active gatekeeper becomes unavailable for any reasons (such as power
failure, scheduled maintenance, or failure to respond to three successive hello packets), the standby
gatekeeper assumes the active role transparently within a few seconds. Because the new active
gatekeeper assumes both the IP and MAC addresses of the phantom, video terminal registrations time
out, and the terminals re-register with their same IP address to the newly active gatekeeper.
Note When configuring gatekeepers and border elements on routers supporting HSRP, configure the border
elements to register with the virtual, or phantom, IP address of the gatekeeper pair. This configuration
enables both border elements to register with the active gatekeeper so that video calls are load-balanced
between the two devices. If the primary router fails, the border element on the standby router registers
with the now active gatekeeper, and calls are forward through it. The border element configured on the
primary router will not re-register with the standby router if the primary router fails.
Note Gatekeeper clustering is not supported in a videoconferencing environment. For clustering to work,
video endpoints would have to support alternate gatekeepers, but currently there are no video terminals
with this support.
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Cisco Unified Border Element
Figure 8-10 Network with Two Directory Gatekeepers
Border Element
The Cisco Unified Border Element has the ability to proxy call signaling and media by termination and
re-origination. The border element can register with the gatekeeper and be inserted in the call flows. The
Cisco Unified Border Element has the following functionality:
Network isolation is achieved by re-origination of signaling and media for the call. You can use
multiple interfaces to provide enhanced isolation.
Classification of signaling, audio, and video traffic can be done with IP Precedence, DSCP, and
Resource Reservation Protocol (RSVP).
Call admission control with the help of the Resource Availability Indicator (RAI) as a gateway
function is widely used. In addition, call treatment based on CPU, memory, total calls, circuit ID
capacity, and gatekeeper bandwidth can be used in various combinations to provide efficient call
admission control. RSVP can be used for dynamic call admission control.
Topology hiding is used for inter-working various types of signaling and protocol.
The border element can provide ease of deployment with firewalls and NAT because it can be a
single trusted entity in either the inside network or the DMZ. With its protocol awareness for NAT
and firewalls, the border element can also reduce management overhead.
Note Because the gatekeeper is an optional device, the border element can also be used as a standalone device.
The border element will have call routing management overhead in this case.
Figure 8-11 illustrates a call across a WAN link through border elements.
7
4
6
8
0
Site A Site B
Site G
Site E
Branch Branch
Regional
E has routes for
E, A and B
Headquarters
Branch Branch
A has routes
for A only
B has routes
for B only
C has routes
for C only
D has routes
for D only
Regional
F has routes for
F, C and D
G has routes for
G, E, F, A, B, C and D
Site C Site D
Site F
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Chapter 8 Cisco Video Infrastructure Components
Collaboration for Desktop Sharing
Figure 8-11 Call Across a WAN Through Border Elements
Note Single-legged Cisco Unified Border Element is supported in Cisco IOS Release 12.3(15)T or later.
Collaboration for Desktop Sharing
Deployments that want to leverage collaboration in their video networks can use any of the following
methods to share user desktop screens.
H.239-Based
With this method, an additional video channel is established by the endpoints. The desktop screen is sent
using this additional video channel during the call. In this method, the endpoint capabilities determine
the screen sharing resolution and characteristics. All participants must support this mechanism. Audio
and video for the conference is through regular channels by the endpoints, as in point-to-point calls.
The following design considerations apply to this method:
Endpoints must support H.239.
Intermediate devices such as call agents, border elements, NAT devices, or firewalls must support
H.239.
Additional WAN bandwidth might be needed and must be provisioned and accounted for.
Server and Client Communication
This method is commonly used by collaboration systems. Cisco Unified MeetingPlace, WebEx, and
other systems use a similar method. In this method, a user-side client is used that can run on a PC or
laptop and that makes desktop screen sharing possible. The server is the application server providing the
services to the clients. Users that do not have access to a PC cannot see the screen being shared. Audio
and video for the conference is through regular channels by the endpoints, as in point-to-point calls.
The following design considerations apply to this method:
Deployments need to consider the operating system and browsers for the user client on the desktop
or the laptops.
Consider the way the client software can be deployed for internal and external devices. Most systems
will check installations or upgrades before users join the collaboration sessions through browsers.
Intermediate devices such as NAT devices or firewalls must support traversal for the user-side client,
if applicable.
H.323 Video
terminal 1
H.323 Video
terminal 2
Gatekeeper
proxy
Leg 1- Terminated call from
video terminal 1 to proxy
Leg 2 - Qos-Enabled call
from proxy to proxy with
IP precedence 4 or RSVP
Leg 3 - Terminated call from
proxy to video terminal 2
Gatekeeper
proxy
8
1
0
5
4
QoS Enabled
IP WAN
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Collaboration for Desktop Sharing
Quality of Service classification and call admission control are required for calls.
Additional WAN bandwidth might be needed and should be accounted for and provisioned to
accommodate the collaboration traffic.
Cisco Unified Videoconferencing Desktop
The Cisco Unified Videoconferencing Desktop solution is similar to the server and client
communication. (See Figure 8-12.)The client component takes care of the video, audio, and desktop
sharing capability. The server component is the application server for collaboration. In addition to this,
it interacts with the MCU that hosts the video conference and enables traditional endpoints in the
conference to view the content shared through the desktop. On the client side it uses a web protocol such
as HTTP/HTTPs to tunnel the data, and on the endpoint side it uses H.239 for screen sharing. In this
way, the Cisco Unified Videoconferencing Desktop provides a combination of H.239 and server/client
communication for the collaboration solution.
Figure 8-12 Cisco Unified Videoconferencing Desktop
The following design considerations apply to this method:
Video endpoints must support H.239.
Intermediate devices such as call agents, border elements, NAT devices, or firewalls must support
H.239.
Deployments need to consider the operating system and browsers for the user client on the desktop
or the laptops.
Consider the way the client software can be deployed for internal and external devices. Most systems
will check installations or upgrades before users join the collaboration sessions through browsers.
Quality of Service classification and call admission control are required for calls.
Intermediate devices such as NAT devices or firewalls must support traversal for the Cisco Unified
Videoconferencing Desktop client, as applicable.
Additional WAN bandwidth might be needed and should be accounted for and provisioned to
accommodate the collaboration traffic.
MCU
2
5
3
1
9
3
H.323 Room
System
Cisco Unified
Videoconferencing
Desktop
Cisco Unified
Videoconferencing
Desktop Server
Cisco Unified
Videoconferencing
Desktop
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Collaboration for Desktop Sharing
Cisco recommends co-locating the desktop server and the conference MCU because the Cisco
Unified Videoconferencing Desktop server initiates a video channel to the MCU for every desktop
client.
Conference Streaming
Some conferences such as training sessions, large team meetings, or panel discussions have the key
people such as the trainer, team leader, or moderators and panel numbers participate interactively in the
conference, while the large majority of users are viewing participants only. Rather than have every
participant join the conference with their video endpoints, streaming such conferences is a more efficient
and scalable way to present the conference to viewers. (See Figure 8-13.) This not only optimizes the
conferencing resources, but it also allows users to view the conference from simple devices such as
media players that support streaming. For such solutions Real Time Streaming Protocol (RTSP) is
commonly used.
The Cisco Unified Videoconferencing Desktop server supports streaming of live conferences. The
participants who will be transmitting in the conference can join the conference through conventional
endpoints that support interactive calls. The server uses the active conference from the MCU and
converts it into an RTSP stream. The viewers can also view the conference using common players such
as Quicktime player, VLC media player, Windows Media player, Real media player, or any other players
that support RTSP.
Figure 8-13 Conference Streaming
The following design considerations apply to conference streaming:
The streaming media might lag behind actual conference.
Quality of Service classification and policing traffic are required.
Unicast streaming may be used with a small number of viewers.
Multicast streaming can be used if there is a need to conserve WAN bandwidth or if there is a large
number of users viewing conferences.
Additional WAN bandwidth may be needed and should be accounted for and provisioned to
accommodate the streaming traffic.
2
5
3
1
9
4
Cisco MCU
Cisco Unified
Videoconferencing
Desktop
Cisco Unified
Videoconferencing
Desktop Server
H.323 Signaling
RTP Media
RTSP Media
Streaming
Browser Client
Cisco Unified
Videoconferencing
Manager
RTSP
Streams
RTP
Media
Streaming
Browser Client
Streaming
Browser Client
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Chapter 8 Cisco Video Infrastructure Components
Collaboration for Desktop Sharing
Conference Recording
Recording the conference provides users with the capability to view the conference at a later time.
Recording servers are separate application servers that store the recording and make it available for later
viewing. The Cisco Unified Videoconferencing Desktop server supports recording of conferences.
Conferences can be recorded by selecting that option when scheduling the conferences or by the
moderator through the user desktop.
The following design considerations apply to conference recording:
The recording server should be deployed on a separate server. Co-hosting with other Cisco Unified
Videoconferencing server should be done only after carefully considering the server capacity and
scalability.
Recordings must be stored and managed on the desktop recording server. Access can be through the
Recording access webpage. Security of these recordings depends on the operating system of the
recording server.
The recording server capacity should be planned based on the type of recordings, the frequency of
recording, and how long the recordings will be available to users before the system archives them.
Enterprises need to define a conference recording storage and retention policy.
Conference recording is dependent on the recording bitrate configured on the recording server. Table 8-5
provides brief guidelines on the storage needed for recording meetings.
Use the following general formula to calculate the minimum server storage capacity needed for
recording conferences:
Minimum Storage Capacity = [Recorded Mbytes/min (User corresponding values from Table 8-5
based on the recording bitrate)] [Recording time per day in minutes] [Number of days]
[Number of simultaneous conferences]
For example, if the recording rate is 256 kbps and you want to be able to record 15 full days (24 hours)
of 5 simultaneous conferences, the minimum required storage capacity would be:
Minimum Storage Capacity = 2.3 MB/min [60 min/hr 24 hr/day] [15 days] [5 conferences]
Minimum Storage Capacity = 248.4 GBytes
Table 8-5 Storage Guidelines for Recording Servers
Recording Bitrate (kbps) Amount of Data Recorded (MBytes/min)
1
1. These values include a factor of 20% for overhead.
256 2.3
384 3.5
512 4.7
768 7.0
1024 9.2
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Collaboration for Desktop Sharing
Integration and Interoperability
The Cisco Unified Videoconferencing MCU can be used for various system integration needs and to
fulfill interoperability requirements with various systems, as described in the following sections.
Telepresence Integration
The Cisco Unified Videoconferencing 5000 Series or 3500 Series MCUs provide Cisco Telepresence
Multipoint Switch integration to traditional videoconferencing networks. This allows H.323 terminals,
desktop telephony, and other videoconferencing devices to participate in a conference with Cisco
Telepresence systems. The Cisco Telepresence Multipoint Switch extends the conference to the Unified
Videoconferencing MCU. This enables traditional and video telephony devices to join into the MCU
conference, while the Cisco Telepresence system is conferenced through the Cisco Telepresence
Multipoint Switch as a single conference.
For additional details, refer to the Cisco Telepresence Multipoint Switch documentation at
http://www.cisco.com/en/US/products/ps7315/tsd_products_support_series_home.html
WebEx Integration
Through the deployment of the entire Cisco Unified Videoconferencing portfolio, especially the Desktop
Streaming Server, WebEx clients on WebEx site versions T27 or later can be configured to use the
Advanced Video plug-in. Advanced Video enables on-premise room systems, video telephony, and even
telepresence systems to participate in a video conference with webcams both on-premise and over the
Internet.
Integrating the Cisco Unified Videoconferencing MCU of the enterprise with WebEx leverages the Cisco
Unified Videoconferencing Desktop solution to bring in the video of traditional desktop systems and
traditional videoconferencing systems and presenting it in a WebEx collaboration session. In this way,
the Native Video support in WebEx gets replaced with the one from the enterprise MCU. (See
Figure 8-14.)
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Collaboration for Desktop Sharing
Figure 8-14 WebEx Integration
Microsoft Office Communicator
Enterprises that deploy Microsoft Office Communicator need integration for desktop video. The desktop
client uses Cisco Unified Videoconferencing Desktop server integration with Microsoft Office
Communicator to display video from the MCU in Microsoft Office Communicator, while using video
from the desktop cameras in the conference as user video. Scheduling conferences is done through
Outlook Plug-in integration.
Lotus Sametime
Enterprises that deploy Lotus Sametime need integration for desktop video. The desktop client uses
Cisco Unified Videoconferencing Desktop server integration with Sametime to display video from the
MCU, while using video from the desktop cameras in the conference as user video. Scheduling
conferences is done through Lotus Notes Plug-in integration.
For more details on the integrations, refer to the latest Install and Upgrade Guides for the Cisco Unified
Videoconferencing Desktop, available at
http://www.cisco.com/en/US/products/ps7088/prod_installation_guides_list.html
2
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WebEx Data Center Enterprise Network
Standards-based
video endpoint
Cisco MCU
Cisco Unified
Videoconferencing
Desktop Server
H.323/SIP
MCU conference
video through
the Desktop
Server to
Desktop client
WebEx Client with Cisco
Unified Videoconferencing
Desktop integration
WebEx
Cconference
traffic
WebEx Client with Cisco
Unified Videoconferencing
Desktop integration
External Network
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Collaboration for Desktop Sharing
Connecting the Video Network with External Untrusted Networks
Enterprises want to leverage the benefits of videoconferencing not only for their internal users but also
with external enterprises and users. The following design considerations can help identify the best
approach.
External Connectivity
External connectivity to the enterprise network is a key requirement. The external devices can also be
devices at the enterprise partners, vendors, or suppliers within a closed user group (CUG), but they might
be untrusted by the enterprise. In some cases the external devices may be devices on the networks of
other enterprises or on the untrusted Internet.
Note Quality of Service for calls might require special consideration because QoS on the Internet is best-effort
and cannot be guaranteed. Enterprise QoS can address quality within the enterprise network. For
additional details, refer to the Enterprise QoS Solution Reference Network Design Guide, available at
http://www.cisco.com/go/designzone.
Dial Plan
A key consideration for external connectivity is how the enterprise routes the calls. Does the enterprise
act as a peer with other enterprises to route calls between themselves through call agents, gatekeepers,
or SIP proxies, based on a protocol of choice?
Enterprises can also assign E.164 addresses to external endpoints so that calls can be routed only within
the enterprise network to those external endpoints, thus controlling the dial plan to them. In this case,
you must ensure that external networks cannot use the enterprise-assigned dial plan to reach these
external devices. Enterprises can enforce rules or registration to prevent unauthorized access.
Enterprises that wish to support external devices that dial by IP address for conferences can use
mechanisms that direct callers to a fixed enterprise number and present them to an auto-attendant that
provides the endpoint information to select and join conferences using DTMF digits. Such deployments
can leverage the Cisco Unified Border Element, which can service calls that do not have a called number
in the call setup and can present that call with an auto-attendant number as the called number to an MCU
with auto-attendant service.
External Endpoints or Devices
Identifying the type of endpoint can significantly change the way connectivity to external networks can
be achieved.
Traditional endpoints or desktop endpoints support standards-based protocol such as H.323 or SIP and
can register with the gatekeeper or SIP proxy as applicable. These endpoints then can register with
internal trusted devices. Cisco recommends firewalls that support application-level gateway inspection.
Topology hiding devices can provide a more efficient method for deployment and management.
Desktops or laptops with no clients might not support standards-based protocols but might use tunneling
through HTTP/HTTPs to the communication servers. To service such clients, the Cisco Unified
Videoconferencing Server can be deployed in the DMZ network. Users can access the server using the
Microsoft Internet Explorer (IE) browser and can authenticate through credentials and the conference
personal identification number (PIN) to participate in the conference. The lightweight software for the
desktop or the PC can be downloaded and installed if needed.
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Firewalls and Network Address Translation (NAT)
Firewalls and Network Address Translation (NAT)
Firewalls provide security for unauthorized access to internal trusted networks. Cisco recommends that
the most secure way to prevent unauthorized access for VoIP traffic is by inspection of the VoIP
registrations and calls. Enterprises may need to deploy application-aware firewalls. These firewalls can
then inspect on VoIP protocols too.
H.323 uses multiple signaling ports. An application-aware firewall can inspect the H.323 messages and
open respective ports for the call to proceed. For an H.323 call to take place, it must first open an H.225
connection on TCP port 1720 using Q.931 signaling. Next, the H.245 management session is established.
While this session can take place on a separate channel from the H.225 setup, it can also be done using
H.245 tunneling, which takes the H.245 messages and embeds them in the Q.931 messages in the
previously established H.225 channel. Next, the H.245 session opens dynamically assigned ports for the
UDP-based RTP and RTCP video and audio data streams. The port numbers can range from 1024 to
65535. Because the port numbers are not known in advance, and because it would defeat the purpose of
a firewall to open all these ports, a firewall must be able to snoop the H.323 data stream in order to open
the additional ports needed for the call. This snooping is also known as stateful inspection. Firewalls that
support H.323 message inspection in order to open just the needed ports per call are termed
application-aware firewalls.
An additional problem encountered with most firewalls is the use of Network Address Translation
(NAT). Within H.323, the H.225 and H.245 signaling channels make heavy use of the embedded IP
address. For example, assume a terminal has a private address of 10.1.1.125, which gets translated to
206.165.202.125 when it tries to place a call to an H.323 terminal with an IP address of 206.165.201.78
on the outside network. The terminal on the outside still receives the private address within the H.225
signaling stream. Because this is a non-routable address, an attempt to make a connection back will fail.
One way to work around this problem is to use an H.323-aware NAT firewall, which can rewrite the
addresses in the signaling payload.
Using the border element in NAT or firewall environments allows administrators to target a single IP
address to terminate all H.323 video calls. All incoming and outgoing video calls that access the public
network will use the border element. With the Cisco ASA Firewall, administrators can enable H.323
fix-up and allow UDP port 1720 traffic to access the IP address of the border element. Without the border
element, administrators would have to configure UDP port 1720 to all videoconferencing devices and
have static NAT for each device, which may not be scalable with a large number of video endpoints in
the network. If you use a Cisco IP/VC 352x or 3540 gateway, port 1820 must be configured for the
videoconferencing devices. Figure 8-15 illustrates the call flow in a network with NAT and a firewall.
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Chapter 8 Cisco Video Infrastructure Components
Firewalls and Network Address Translation (NAT)
Figure 8-15 Call Flow with NAT and a Firewall
When using the gatekeeper on the border element, you can have external devices on the outside untrusted
network register to the gatekeeper through the UDP 1718 or 1719 port pinhole to the gatekeeper for RAS
registrations. The application-aware firewall inspects RAS signaling and opens pinholes for H.225 call
signaling based on the embedded ports in the RAS messages. Additional inspection of H.225 can open
ports dynamically for H.245 communication and then RTP media, respectively, through the
application-aware firewall. Organizations can use external gatekeepers to ensure E.164 number
resolution.
SBC
WAN
IP PSTN
Internet
2
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SBC
V
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Firewalls and Network Address Translation (NAT)
C H A P T E R
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9
Multi-Zone WAN Case Study
Last revised on: October 30, 2009
This chapter provides an example of a typical WAN multi-zone model deployed in an enterprise
environment.
In this case study, the enterprise is a health care provider with locations spread across the United States.
Five locations currently use ISDN-based videoconferencing. The enterprise has a T1 to each site and
would like to install new H.323 videoconferencing units and utilize their existing WAN bandwidth. Each
site contains a minimum of three video units, and the enterprise has standardized on 384 kbps as their
call data rate. The enterprise requires multipoint calls as well as the ability to call off-net to their clients.
Network Topology
Currently the enterprise in this example has five sites in the United States, consisting of Sacramento CA,
Los Angeles CA, Dallas TX, Columbus OH, and Chicago IL. Each site connects back to Columbus, the
headquarters, with a T1 link. The bandwidth utilization on all the connections is low. The enterprise has
just upgraded their WAN routers at remote sites to Cisco 2851 routers to support voice, video, and data
in the near future. The headquarters is connected with a T1 line to the internet. Currently, all
videoconferencing units are directly connected to an IMUX with three BRI lines, allowing bonded
384-kbps calls. The Columbus site contains an H.320 multipoint control unit (MCU) with three PRI lines
supporting multipoint calls among the sites. Some remote specialists use a public gatekeeper on the
internet to reach the health care provider. Figure 9-1 illustrates the current IP network, and Figure 9-2
illustrates the current videoconferencing network.
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Chapter 9 Multi-Zone WAN Case Study
Network Topology
Figure 9-1 Current IP Network
Figure 9-2 Current Videoconferencing Network
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Si
Si
Si
Si
Si
Si
Si Si
Cisco 3640
Chicago IL,
single building
Si
Si
Cisco 3640
Dallas Tx, 3 Bldg. campus
with 100 Mbps connections
between buildings
Columbus OH,
7 Bldg. campus
with 100 Mbps
connections
between buildings
Si
Si
Cisco 3640
Los Angeles CA
single building
Leased
T1 lines
Si Si
Cisco 3640
Cisco
7200
Sacramento CA, 2 Blkg. campus
with 100 Mbps connections
between buildings
7
4
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5
PSTN
ISDN
Columbus OH,
24 H.320 videoconferencing units directly connected
to the ISDN network with 3 BRI lines each. One H.320
MCU connected to the ISDN network with 3 PRI lines.
Sacramento CA,
6 H.320 videoconferencing units
directly connected to the ISDN
network with 3 BRI lines each.
Los Angeles CA,
4 H.320 videoconferencing units
directly connected to the ISDN
network with 3 BRI lines each.
Dallas TX,
10 H.320 videoconferencing units
directly connected to the ISDN
network with 3 BRI lines each.
Chicago IL,
3 H.320 videoconferencing units
directly connected to the ISDN
network with 3 BRI lines each.
BRI
BRI
BRI
BRI
BRI
PRI
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Chapter 9 Multi-Zone WAN Case Study
Network Design
Network Design
The network outlined in the previous section is a classic WAN multi-zone model. There is sufficient
WAN bandwidth, and each site contains three or more video terminals. In this network, a gatekeeper and
border element are located at each site. Directory gatekeeper services are configured, and Hot Standby
Routing Protocol (HSRP) is used for gatekeeper redundancy at the Columbus site. Quality of service
(QoS) and call admission control (CAC) need to be configured in the network to ensure video quality.
Firewall security is used to connect to the internet, and the border element is the trusted device through
the firewall.
Quality of Service (QoS)
End-to-end QoS is a key factor in a successful deployment. The enterprise in this example has decided
to use an H.323 video terminal that supports marking of IP Precedence. The Columbus, Sacramento, and
Dallas sites have upgraded to Catalyst 6500 switches. In these three sites, LAN QoS will be configured;
the remaining three sites will support LAN QoS when the switches at those sites are also upgraded. All
video units will be connected to 10/100 Ethernet ports.
All video terminals, gateways, and MCUs are configured to mark IP Precedence 4. In Columbus,
Sacramento, and Dallas, trust boundaries are set on the Catalyst 6500 switches. Video gateways and
MCUs are also installed in Columbus, Sacramento, and Dallas. For gateways and MCUs that do not
support IP Precedence, IP Precedence is marked and a trust boundary is set on the Catalyst 6500 ports
to which the gateways and MCUs are connected. Gateways are also installed in Los Angeles and
Chicago.
Priority queues are configured on all WAN routers and are provisioned for 920 kbps. This guarantees
that bandwidth is available for two 384-kbps calls. An access list entry is also added on the WAN router
to set the entrance criterion for the priority queue. Only video traffic received from the border element
is admitted to the priority queue.
The gatekeeper at each site is configured to use the local border element for all inter-zone calls. The
border element rewrites IP Precedence 4 and provides a single access point to the configured priority
queue.
For more information regarding network QoS, refer to the Enterprise QoS Solution Reference Network
Design Guide, available at http://www.cisco.com/go/designzone.
Call Admission Control
Call admission control (CAC) must be implemented for inter-zone calls, and it is also a good idea to
configure CAC for intra-zone calls. Enabling CAC for inter-zone calls guarantees that the bandwidth
limits provisioned on the priority queues are not exceeded. If the provisioned bandwidth for the priority
queue on the WAN route is exceeded, all video calls in the queue will be affected. The gatekeeper at
each site contains the following three bandwidth statements for CAC.
bandwidth total {default | zone <zone-name>} <bandwidth-size>
bandwidth remote 1536
bandwidth session default 768
It is important to note that the bandwidth is calculated in half-duplex mode, so the call data rate must be
doubled. A 384-kbps call is represented as 768 kbps in the bandwidth statements. The bandwidth total
command allows administrators to limit the bandwidth within a single local zone, or for all local zones
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Chapter 9 Multi-Zone WAN Case Study
Network Design
by adding the default statement. The remote bandwidth (available bandwidth to and from any remote
zone) is limited to 1536 kbps, or two 384-kbps calls. The bandwidth per session is limited to 768 kbps,
or one 384-kbps call.
Figure 9-3 illustrates QoS and CAC points for Columbus, Figure 9-4 illustrates QoS and CAC points for
Dallas and Sacramento, and Figure 9-5 illustrates QoS and CAC points for Los Angeles and Chicago.
Figure 9-3 QoS and CAC for Columbus
Figure 9-4 QoS and CAC for Dallas and Sacramento
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2 Cisco 3620 routers providing gatekeeper and
proxy functionality and configured with HSRP.
CAC configured on gatekeepers and IP
Precedence set to 4 on the proxy.
CAC set for intra-zone
and inter-zone calls.
Video
terminal
Trust
boundaries
configured
MCU or
gateway
IP precedence
set to 4
IP precedence set
to 4 and trust
boundary configured
ACL added to set PQ entrance
Criterion. Only packets from the
proxy will have access to the PQ.
Priority Queue (PQ)
configured and
provisioned for 920 kbps
on each T1.
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Video
terminal
Trust
boundaries
configured
MCU or
gateway
IP precedence
set to 4
WAN
router
IP precedence set
to 4 and trust
boundary configured
ACL added to set PQ entrance
Criterion. Only packets from the
proxy will have access to the PQ.
Gatekeeper and proxy functionality configured
on WAN router. CAC configured on gatekeeper
and IP precedence set to 4 on the proxy.
Priority Queue (PQ) configured
and provisioned for 920k
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Dial Plan
Figure 9-5 QoS and CAC for Los Angeles and Chicago
Dial Plan
When deciding on a dial plan, it is always a good idea to start with the incoming PSTN call routing. In
our example, we have created five zones that all contain video gateways. DID is used to route incoming
calls to video terminals. IVR is used to route calls from the video gateways in Columbus, Dallas, and
Sacramento, to their local MCUs. If, for some reason, one or more of the zones in our example did not
contain a gateway, IVR for routing all incoming PSTN calls would have been a better choice.
Zone Prefixes
The zone prefix for each zone is based on the local area code. Area codes are unique, and users are
familiar with the numbering structure. In our configuration there is a single zone in each site, so the zone
prefixes are based on area codes. If more than one zone were required in a single area code, longer zone
prefixes could be used. (Refer to Zone Prefix Design, page 6-7.) The zone prefixes in this network are:
Columbus = 614
Sacramento = 916
Dallas = 972
Chicago = 847
Los Angeles = 213
Service Prefixes
Service prefixes must be configured for all MCUs and video gateways. As described in the chapter on
Dial Plan Architecture, it is a good idea to reserve a block of numbers for video gateways and a block of
numbers for MCUs. In this case, the enterprise has chosen to standardize on 384-kbps calls; this makes
service prefixes for gateways very simple. The obvious choice would be to use 9 for all PSTN calls, but
that would cause routing problems in the Sacramento and Dallas zones. The Sacramento zone prefix is
916, and overlapping gateway service prefixes and zone prefixes will cause routing problems. There are
7
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Video
terminal
Video
gateway
IP precedence
set to 4
WAN
router
ACL added to set PQ entrance
Criterion. Only packets from the
border element will have access to the PQ.
Gatekeeper and border element functionality configured
on WAN router. CAC configured on gatekeeper
and IP precedence set to 4 on the border element.
Priority Queue (PQ) configured
and provisioned for 920k
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Chapter 9 Multi-Zone WAN Case Study
Dial Plan
two options; reserve another block of numbers other than 9*, or use a service prefix such as 9#. In this
case, we have chosen 9# for PSTN access in all zones. Any time a user tries to access the WAN, the dial
string will start with 9#.
For MCU service prefixes, 8* is reserved, and the zone prefix is appended to associate it with the zone
in which the MCU resides. The MCU service prefix in the Sacramento zone is 9168*, and in Los Angeles
it is 2128*. Table 9-1 lists the service prefixes chosen for different types of calls on the MCU. (These
service prefixes are used in every zone, and the zone prefix is appended.)
E.164 Addresses and H.323-IDs
The carrier provides E.164 addresses for this enterprise. Because DID has been chosen for incoming
PSTN routing method, the enterprise will order blocks of DID numbers with each PRI line. Each video
terminal is assigned a valid DID number for its E.164 address. In Columbus, there are 24 video
terminals, and thirty DID numbers are ordered with the PRI line for Columbus. (The extra six numbers
are for expansion.) In Los Angeles and Chicago, DID numbers off the BRI lines will be used as E.164
addresses. (See Video Infrastructure, page 9-7, for the video components at each site.)
H.323-IDs are based on the name of the conference room where the video system resides. Since the video
terminals may be moved from room to room, H.323-IDs will not be used for dialing. The enterprise is
using a global address book that will display all of the IP video terminals on the network. Users can
choose to dial from the address book or manually enter the E.164 address of the unit being called.
Figure 9-6 illustrates the Dial plan in Columbus.
Table 9-1 MCU Service Prefixes
Service Prefix Data Rate
Number of
Parties Video Format Continuous Presence
83 384 kbps 3 H.261 No
84 384 kbps 4 H.261 Yes
85 384 kbps 5 H.261 No
89 384 kbps 9 H.261 No
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Chapter 9 Multi-Zone WAN Case Study
Video Infrastructure
Figure 9-6 Dial Plan for Columbus
Video Infrastructure
When deciding on location and number of video components, it is important to understand the
enterprise's needs. This enterprise made it clear that less than ten percent of all video calls were off-net
calls. The number of video calls placed daily ranges from 10 to 15, and most calls are multipoint. For
this reason, the enterprise decided to go with video gateways at each site and MCUs in Columbus,
Dallas, and Sacramento. The following sections list the video components for each site. All IP video
calls for the enterprise are received at the headquarters.
Columbus
IP Video Terminals, 24
The current 24 H.320 video systems are over three years old and will be replaced with new H.323
group systems that support IP Precedence.
MCUs, 4
The four MCUs will be configured in a stack, allowing one set of service prefixes to be shared by
all four MCUs.
Video Gateway PRI, 1
A single PRI gateway will be installed with 30 DID numbers that will be assigned to the IP video
terminals. All 10 digits will be passed to the gateway by the carrier, allowing each video terminal
to register with a 10-digit fully qualified E.164 address. IVR will be enabled and used for PSTN
access to MCU conferences. One DID number will have to be reserved for IVR calls.
7
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Zone Prefix: 614
Video Terminal
E.164 Address
6143251010
H323-ID
Indians
MCU Service Prefixes:
61483
61484
61485
61486
Video Gateway
Service Prefix: 9#
DID Range: 6143251000-6143251030
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Video Infrastructure
Sacramento
IP Video Terminals, 6
The existing six H.320 video systems are over three years old and will be replaced with new H.323
group systems that support IP Precedence.
MCU, 1
A single MCU will be located on the Sacramento campus for local, on-site multipoint calls. The
Sacramento campus is in the process of adding another building and possibly adding two or three
additional IP video terminals. The MCU will also allow multiple video terminals to participate in
an off-campus multipoint call while consuming the bandwidth of only a single call. This will be
done by cascading a Sacramento MCU conference with a Columbus MCU conference.
Video Gateway, 1
A single PRI gateway will be installed with 10 DID numbers that will be assigned to the IP video
terminals. All 10 digits will be passed to the gateway by the carrier, allowing each video terminal
to register with a 10-digit fully qualified E.164 address. IVR will also be enabled and used for PSTN
access to MCU conferences. One DID number will have to be reserved for IVR calls.
Dallas
IP Video Terminals, 10
The existing 10 H.320 video systems are over three years old and will be replaced with new H.323
group systems that support IP Precedence.
MCU, 1
A single MCU will be located on the Dallas campus for local, on-site multipoint calls. The MCU
will also allow multiple video terminals to participate in an off-campus multipoint call while
consuming the bandwidth of only a single call. This will be done by cascading a Dallas MCU
conference with a Columbus MCU conference.
Video Gateway, 1
A single PRI gateway will be installed with 15 DID numbers that will be assigned to the IP video
terminals. All 10 digits will be passed to the gateway by the carrier, allowing each video terminal
to register with a 10-digit fully qualified E.164 address. IVR will also be enabled and used for PSTN
access to MCU conferences. One DID number will have to be reserved for IVR calls.
Los Angeles
IP Video Terminals, 4
The existing four H.320 video systems are over three years old and will be replaced with new H.323
group systems that support IP Precedence.
MCU, 0
Los Angles will not have a local MCU.
Video Gateway, 1
A single BRI gateway will be installed with four BRI lines. Each video terminal will receive a DID
number from one of the BRI lines. IVR will not be enabled on the gateway.
Chicago
IP Video Terminals, 3
The existing three H.320 video systems are over three years old and will be replaced with new H.323
group systems that support IP Precedence.
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Video Infrastructure
MCU, 0
Chicago will not have a local MCU.
Video Gateway, 1
A single BRI gateway will be installed with three BRI lines. Each video terminal will receive a DID
number from one of the BRI lines. IVR will not be enabled on the gateway.
Figure 9-7 illustrates the video components and dial plan for the new IP video network.
Figure 9-7 Dial Plan for Example Video Network
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Gateway
IP/VC 3525
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Prefix 9#
Gateway IP/VC 3525
Service Prefix 9#
Gatekeeper
proxy
24-IP Video terminals
E.164 address range
614325100-6143251023
6-IP Video terminals
E.164 address range
9162252010-
9162252015
MCUs
Service
Prefix
61483
61484
61485
61489
MCU Service Prefix
61483
61484
61485
61489
Sacramento
zone 916
Columbus
zone 614
Si
Gateway IP/VC 3525
Service Prefix 9#
Gatekeeper
proxy
10-IP Video terminals
E.164 address range
9724242350-
9724242459
MCU Service Prefix
97283
97284
97285
97289
Dalas
zone 972
Gateway IP/VC 3520
Service Prefix 9#
Gatekeeper
proxy
4-IP Video terminals
E.164 address range
2135242052-
2135242055
Los angeles
zone 213
Gateway IP/VC 3520
Service Prefix 9#
Gatekeeper
proxy
3-IP Video terminals
E.164 address range
8477621356-
8477621358
Chicago
zone 847
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Chapter 9 Multi-Zone WAN Case Study
Video Infrastructure
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A P P E N D I X A
Interworking with Session Initiation Protocol
(SIP)
Last revised on: October 30, 2009
The most popular protocol used by enterprises for video conferencing is H.323. That protocol has been
used for a long time, and capabilities such as H.235 for security and H.239 for data sharing make it
popular. However, newer endpoints and systems support SIP too, which is seeing increased adoption.
Enterprises that want to use a common protocol for voice, video, instant messaging, and presence would
like to standardized on SIP, and new deployments with call agents that have SIP servers providing call
resolution find the support of SIP in video communication devices desirable. Enterprises that use Secure
RTP (sRTP) for call encryption and do not need data sharing capabilities through the call, may prefer to
deploy endpoints with SIP protocol.
A SIP network can consist of various SIP servers that provide the following functionality:
Registration
The SIP registrar server provides registration for the endpoints. Usually the SIP server that provides
the dial plan capabilities would also support the registration functionality.
Signal proxy
SIP proxy servers proxy the call signaling through them. The proxy servers can support two modes:
Proxy the initial call signaling and then have the SIP servers control the call.
Proxy the call and the signaling on behalf of the SIP server. In this mode the SIP proxy operates
in a record route mode.
Location services.
The location services inform the SIP registrar of a new location of an endpoint. In scalable networks,
this is useful to associate properties of that location with the endpoint.
A typical SIP network consists of all these functions in a single server, or multiple servers can be used
with the functionality distributed to each server according to its role.
Note Call agents such as Cisco Unified Communications Manager support registration of SIP endpoints and
connectivity with SIP trunks. Cisco Unified Communications Manager supports additional protocols
such as H.323 and Skinny Client Control Protocol (SCCP), and it can also be used to provide
interworking between H.323 networks and SIP networks while supporting large-scale deployments.
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Appendix A Interworking with Session Initiation Protocol (SIP)
Give adequate consideration to the MCUs, gateways, and other systems or servers that support SIP
protocol to service all the deployment requirements, so that your system design can support all the
required calls, conferences, and collaboration services needed by your enterprise.
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G L O S S A R Y
Last revised on: October 30, 2009
Numerics
802.1P
802.1Q
802.1P and 802.1Q are the standards proposed by the inter-working task groups of the 802 standards
committee. 802.1Q is the IEEE Standard for Local and Metropolitan Area Networks: Virtual Bridge
Local Area Network (VLAN). 802.1P is the IEEE Standard for Local and Metropolitan Area
Networks Supplement to Media Access Control (MAC) Bridges: Traffic Expending and Dynamic
Multicasting Filtering.
A
ARQ Admission Request
B
Border Element A device that is used in the periphery of the network to separate two different networks and to serve as
a demarcation device
BRI Basic Rate Interface
C
CAC Call admission control
Cascade The process of connecting two or more MCUs to create a larger conference.
Codec Coder-decoder, for digitizing voice and video. Compression algorithms can also be used during the
digitizing process.
CoS Class of Service
cRTP Compressed Real-time Transport Protocol
Glossary
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D
DID Direct Inward Dialing
DSCP Differentiated Services Code Point is an Internet Engineering Task Force (IETF) standard that uses six
bits in the ToS (Type of Service) field of the IPv4 header to specify class of service for each packet.
DTMF Dual Tone Multi-Frequency
E
E.164 Address format used for H.323 devices
EMP Enhanced Media Processor
G
Gatekeeper Used for H.323 registration, call routing, and admission control
G.711 G.711 pulse code modulation (PCM) encoding provides 64 kbps analog-to-digital conversion using
mu-law or a-law.
H
H.261 Video codec protocol
H.263 Video codec protocol
H.323 Standard protocol for audio, video, and data communications across IP-based networks
H.323-ID Alphanumeric identifier assigned to an H.323 video terminal
hopoff Command added to a Cisco gatekeeper for static inter-zone routing
HSRP Hot Standby Routing Protocol
I
IMUX Inverse Multiplexer
IP Internet Protocol
IP Precedence IP Precedence uses the three precedence bits in the ToS (Type of Service) field of the IPv4 header to
specify class of service for each packet.
Glossary
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ISDN Integrated Services Digital Network
IVR Interactive Voice Response
L
LAN Local Area Network
LLQ Low Latency Queuing is a QoS mechanism that ensures the timely queuing of critical, delay-sensitive
traffic.
LRQ Location Request
M
MC Multipoint Controller
MCM Multimedia Conference Manager
MCU Multipoint Control Unit, used for video conferences containing more than two endpoints
MP Multipoint Processor
MSN Multiple Subscriber Number
P
PRI Primary Rate Interface
Proxy H.323-to-H.323 gateway used for assigning QoS and security access
PSTN Public Switched Telephone Network
Q
QoS Quality of Service
R
RAS Registration, Admission, and Status protocol
RRQ Registration Request
RSVP Resource Reservation Protocol
Glossary
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RTCP Real-time Control Protocol
RTP Real-time Transport Protocol
S
SCCP Skinny Client Control Protocol
Service Prefix A digit string used to identify a service on an MCU or gateway
Stacking Grouping MCUs to obtain a larger number of multipoint conferences
T
ToS Type of Service
W
WAN Wide Area Network
WRED Weighted Random Early Detection
WRR Weighted Round Robin
Z
Zone A logical group of H.323 infrastructure components managed by a single gatekeeper
Zone Prefix A digit string used to identify a group of H.323 devices
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I N D E X
Symbols
* (wildcard) 6-3, 6-7, 7-11
Numerics
3522 BRI gateway 8-15
3527 PRI gateway 8-15
3545 gateway 8-15
802.1Q 3-4
A
access control list (ACL) 4-4, 4-7
ACL
multi-zone WAN 4-7
single-zone WAN 4-4
address, E.164 6-2, 7-5, 7-8
address resolution by gatekeeper 7-13
Admission Request (ARQ) 7-13
alternate gatekeeper 8-14, 8-18
area code 6-2
ARQ 7-13
B
balancing the network load 8-16
bandwidth
control 4-6, 5-5
minimum requirements 5-5
provisioning 5-2
bandwidth remote command 4-6
best practices 5-5
border element
described 1-7
functions 8-17, 8-20
gatekeeper function 8-17
usage 5-4
boundary, trust 3-5
C
call admission control
methods 5-5
multi-zone WAN 4-6
single-zone WAN 4-4
call initiation with an MCU 8-4
call routing
dial strings 7-2, 7-3, 7-4
directory gatekeeper 7-9
hopoff statements 7-8
inter-zone calls 7-8, 7-9
methods 7-1
multi-zone example 7-2
PSTN calls in a multi-zone network 7-7
PSTN calls in a single-zone network 7-5
PSTN calls to H.323 7-4
scalability 8-19
scenarios 7-1
campus
infrastructure 3-1
multi-zone deployment model 2-4
QoS 3-4
single-zone deployment model 2-2
traffic classification 3-4
capabilities of partner products 7-6
Index
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capacity planning 5-2
cascading MCUs 8-5, 8-6
case study 9-1
CBWFQ 4-4
Cisco Unified Border Element 1-7, 8-17, 8-20
Cisco Unified Videoconferencing 3522 BRI
Gateway 8-15
Cisco Unified Videoconferencing 3527 PRI
Gateway 8-15
Cisco Unified Videoconferencing 3545 Gateway 8-15
Cisco Unified Videoconferencing Manager 8-8, 8-10
class-based weighted fair queuing (CBWFQ) 4-4
classification of traffic 3-4, 4-3, 4-6, 5-3
class of service (CoS) 3-4
clustering of gatekeepers 8-14
components
dial plan 6-2
videoconferencing 1-3, 8-1
composite deployment model 2-1
compressed RTP 5-5
conferences
multipoint 8-2
scheduled 8-7
CoS 3-4
D
default extension 7-5, 7-8
deployment example 9-1
deployment models
composite model 2-1
multi-zone campus 2-4
multi-zone WAN 2-7
overview 2-1
single-zone campus 2-2
single-zone WAN 2-5
dial plan
architecture 6-1
components 6-2
multi-zone 6-8
single-zone 6-5
dial strings 6-2, 7-2, 7-3, 7-4
DID 7-4, 7-5, 7-7
Differentiated Services Code Point (DSCP) 3-4, 5-3, 5-4
Direct Inward Dialing (DID) 7-4, 7-5, 7-7
directory gatekeeper 7-9
distributed MCUs 8-5
DSCP 3-4, 5-3, 5-4
dynamic cascading of MCUs 8-6
E
E.164 address
default 7-5, 7-8
defined 6-2
entrance criteria 4-4, 4-7
example deployment 9-1
extension, default 7-5, 7-8
F
FIFO 5-4
firewall 8-28
first-in-first-out (FIFO) 5-4
G
gatekeeper
address resolution 7-13
alternate 8-14, 8-18
call admission control 5-5
clustering 8-14
directory 7-9
integration with Resource Manager 8-13
overview 1-4
redundancy 8-18
registration of endpoints 8-12
Index
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requirements and recommendations 8-17
Gatekeeper Update Protocol (Cisco GUP) 8-18
gateway
overview 1-5
service prefix 6-4, 8-16
video 8-15
glossary GL-1
GUP 8-18
H
H.225 1-1
H.245 1-1
H.323
basics 1-1
firewall 8-28
ID 6-2, 6-5, 6-9
overview 1-1
standard 1-1
videoconferencing 1-2
videoconferencing components 1-3
H.323-ID 6-2, 6-5, 6-9
hardware components 8-1
hopoff statements 7-8
Hot Standby Router Protocol (HSRP) 8-18, 8-19
HSRP 8-18, 8-19
I
infrastructure
campus 3-1
multi-zone campus 3-3
multi-zone WAN 4-4
network 3-1
QoS 3-4
single-zone campus 3-2
single-zone WAN 4-2
video components 8-1
WAN 4-1
initiating a call with an MCU 8-4
Interactive Voice Response (IVR) 7-4, 7-5, 7-7
interoperability with partner products 7-6
inter-zone calls
routing with a directory gatekeeper 7-9
routing with hopoff statements 7-8
IP Precedence 5-3, 5-4
IVR 7-4, 7-5, 7-7
L
Layer 2 3-4
Layer 3 3-4, 5-3
LLQ 5-4
load balancing 8-16
low-latency queuing (LLQ) 5-4
M
MCU
call initiation 8-4
cascading 8-5
described 1-6
distributed 8-5
dynamic cascading 8-6
registration with gatekeeper 8-12
service prefix 6-3
virtual 8-9
minimum bandwidth requirements 5-5
minimum software releases 5-5
models 2-1
MSN 7-4
Multiple Subscriber Numbering (MSN) 7-4
multipoint conferencing 8-2
Multipoint Control Unit (see MCU)
multi-zone
call routing example 7-2
Index
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campus deployment model 2-4
campus infrastructure 3-3
case study 9-1
dial plan 6-8
routing inbound PSTN calls 7-7
WAN deployment model 2-7
WAN infrastructure 4-4
N
NAT 8-28
network
infrastructure 3-1
load balancing 8-16
Network Address Translation (NAT) 8-28
Network Manager 8-14
P
partner product capabilities 7-6
platforms
partner products 7-6
QoS features supported 3-6
video infrastructure components 8-1
PQ
configuring 4-4, 4-7
provisioning 4-4, 4-7, 5-4
size of 4-4
Precedence, IP 5-3, 5-4
prefix
service 6-2, 6-3, 8-16
technology 6-2
zone 6-2, 6-7, 7-9
prioritization of traffic 5-4
priority queue (see PQ) 4-4, 4-7
product capabilities 7-6
provisioning
bandwidth 5-2
priority queue size 4-4, 4-7, 5-4
PSTN
calls in a multi-zone network 7-7
calls in a single-zone network 7-5
calls to H.323 7-4
Public Switched Telephone Network (see PSTN)
Q
QoS
campus 3-4
features 3-6
tools 5-3
traffic classification 3-4, 4-3, 4-6, 5-3
traffic prioritization 5-4
trust boundary 3-5
WAN 5-1
quality of service (see QoS)
queuing
low-latency (LLQ) 5-4
priority 4-4, 5-4
R
recommended software releases 5-5
recommended traffic classifications 3-4
redundancy 8-14, 8-18
registration of MCU 8-12
releases
recommended 5-5
remote bandwidth 4-6
Resource Manager 8-13
Resource Reservation Protocol (RSVP) 5-5
routing calls (see call routing)
RSVP 5-5
RTP, compressed 5-5
Index
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S
scalability of call routing 8-19
scheduled conferencing 8-7
scheduling conferences 8-7
service prefix
defined 6-2
design 6-3
gateway 6-4, 8-16
MCU 6-3
wildcard 6-3
Session Initiation Protocol (SIP) A-1
single-zone
campus deployment model 2-2
campus infrastructure 3-2
dial plan 6-5
routing inbound PSTN calls 7-5
WAN deployment model 2-5
WAN infrastructure 4-2
SIP A-1
size of priority queue 4-4
SNA 5-4
snooping 8-28
software
minimum releases 5-5
stateful inspection 8-28
supported QoS features 3-6
switch platforms 3-6
Systems Network Architecture (SNA) 5-4
T
TCP 5-4
TCS4 7-4, 7-5, 7-8
technology prefix 6-2
telephony dialing comparison 6-2
terminal, video 1-4, 5-5
tools for QoS 5-3
ToS 3-4
traffic classification
campus 3-4
multi-zone WAN 4-6
recommended classifications 3-4
single-zone WAN 4-3
WAN 5-3
traffic prioritization 5-4
traffic types 3-4
Transmission Control Protocol (TCP) 5-4
trust boundary 3-5
type of service (ToS) 3-4
V
Videoconferencing 3522 BRI Gateway 8-15
Videoconferencing 3527 PRI Gateway 8-15
Videoconferencing 3545 Gateway 8-15
videoconferencing components 1-3, 8-1
Videoconferencing Manager 8-8, 8-10
video gateways 8-15
video shceduling 8-7
video terminal
number allowed 5-5
overview 1-4
virtual MCU 8-9
W
WAN
capacity planning 5-2
case study 9-1
infrastructure 4-1
multi-zone case study 9-1
multi-zone deployment model 2-7
multi-zone traffic classification 4-6
QoS 5-1
single-zone deployment model 2-5
single-zone traffic classification 4-3
Index
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traffic classification 5-3
traffic prioritization 5-4
weighted fair queuing (WFQ) 5-4
WFQ 5-4
wildcard
in service prefix 6-3
in zone prefix 6-7, 7-11
Z
zone prefix
defined 6-2
design 6-7
directory gatekeeper 7-9
dot (.) method 6-7
format 6-7, 7-10
wildcard 6-7, 7-11