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1 200-001 Cisco Video Network Devices Exam Study Guide

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200-001

Cisco Video Network Devices Exam

Study Guide

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Certifications

VIVND (Video Network Devices) Exam

Implementing Cisco Video Network Devices (VIVND) 200-001 is the exam associated with the

Cisco Video Network Specialist certification and CCNA Video certification. It assesses the

knowledge and skills needed to perform in networked video environments by traditional analog

Audio/Visual professionals who install and support solutions such as Cisco TelePresence video

conferencing and mobile video applications. Topics include video concepts, room readiness

recommendations, installation of desktop and single-screen systems, and video conferencing

solutions.

About This Study Guide

This Study Guide provides all the information required to pass the 200-001 – Cisco Video

Network Devices Exam. It however, does not represent a complete reference work but is

organized around the specific skills that are tested in the exam. Thus, the information contained

in this Study Guide is specific to the 200-001 and not the entire Cisco Video Network Devices. It

includes the information required to answer questions related to 200-001 that may be asked

during the exam. Topics covered in this Study Guide includes video concepts, endpoint

configuration, troubleshooting and support and conferencing concepts.

Intended Audience

This Study Guide is targeted at a typical candidate who is a Cisco Certified Network Associate

responsible for the installation, troubleshooting and monitoring of video network devices. Day to

day, the candidate typically manages cisco routers and configure them for video support.

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Good luck!

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Table of content

Video Concepts........................................................................................................5

Describe the functional components of video solutions......................................5

Endpoint Configuration.......................................................................................29

Describe video product models.........................................................................29

Describe environment recommendations..........................................................40

Implement desktop endpoints and surveillance cameras..................................52

Describe features and functions........................................................................66

Troubleshooting and Support.............................................................................75

Describe troubleshooting methodologies..........................................................75

Identify endpoint issues....................................................................................79

Collect system information...............................................................................81

Manage configuration.......................................................................................85

Implement key CLI commands.........................................................................91

Monitor events and alerts..................................................................................95

Conferencing Concepts........................................................................................96

Describe multi point control units.....................................................................96

Describe conferencing features.........................................................................98

Describe scheduling vs ad hoc vs on-demand features...................................100

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Video Concepts

Describe the functional components of video solutions

Architectural Overview

As a process, architecture is the activity of designing and constructing buildings and other

physical structures, primarily to provide shelter. A wider definition often includes the design of

the environment, from the macro level of how a building integrates with its surrounding

landscape to the micro level of the construction details and, sometimes, furniture. Architecture in

its broadest sense is the action of designing a complete system that provides a useful service to

the consumer.

As such, Cisco SBA Collaboration is a system that was created using a structured process to

safeguard the stability of voice, video, and web conferencing for business processes and assets.

The system can be broken down into three primary modular, yet interdependent, components for

your organization. They are the network foundation, network services, and user services, which

have a hierarchical interdependency as shown in the following illustration.

Network Foundation

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The key to the architecture is the network foundation. Similar to the concrete foundation of a

building, the network foundation provides a platform on which everything else relies. As a

standalone layer, the network foundation ensures information is sent dependably from one device

and received at another. How this is accomplished is completely abstracted from the average

user; all they know is that when they pick up the phone, they hear a dial tone. When they place a

call and the other person answers, the audio is clear, and they have a normal conversation. It just

works, and they do not have to think about how the call gets from one point to another.

Intelligent infrastructure devices from Cisco—such as switches, routers, gateways, session

border controllers, and wireless access points—are what make this possible in the background.

Network Services

Network services sit on top of the network foundation. Network services are like the doors,

windows, and walls of the building. A building without these components is just a box. Adding

these services turns the infrastructure into a workable structure, providing reliability, security,

and availability of the organization’s assets. Some users are aware of the value that network

services provide, but do not directly interact with those services. An example of this would be

using a business phone from a home office. The user needs to be behind their VPN router, use a

phone proxy service for a hard phone, or use a Secure Sockets Layer (SSL) VPN client for a soft

phone in order to access business resources. The user does not know or care exactly how the

network services operate. As long as they can make their voice and video calls from wherever

they are at the time, the network services layer is working as expected.

User Services

And finally, user services sit on top of the network services. User services are like the utilities of

the building: water, electricity, phone, Internet, and cable TV services. A user needs direct access

to these services all day long. In the morning, the lights turn on, air conditioners cool, televisions

play content, phones ring, and water is available for morning beverages. As the day progresses,

common utilities are what make the building a comfortable place to work. General user services

for a network include business application software, CRM systems, email, and instant

messaging. User services specific to Cisco include unified communications with voice, web, and

video collaboration.

Cisco Medianet

Cisco Medianet technologies are the recommended approach for video and collaboration

deployments. They span across the three layers and extend the network boundary to include the

endpoints. The network works together with the endpoints in order to scale, optimize, and

enhance the performance of collaboration components.

The idea behind this approach comes from the realization that the endpoints and applications are

the place in the network where most information is stored. The endpoints communicate with the

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network, making the network media-aware and armed with important information that you can

use to make intelligent decisions. The endpoints also become network-aware and are able to

request intelligent network services for troubleshooting.

The Media Services Interface (MSI) that is embedded in Cisco endpoints and collaboration

applications enables the medianet functionality. MSI provides a set of APIs that use medianet

network services, and they also send valuable information about the media flows to the network

devices.

If video is critical your business, Cisco Medianet provides you with a framework to help you

simplify deployment, and troubleshoot and manage all of your video applications.

Network Foundation

Most users perceive the network foundation as a simple transport utility to shift data from one

point to another as fast as possible; many sum this up as ―speeds and feeds‖. In reality, the

network affects all traffic flows and must be aware of end-user requirements and the services

offered. Even with unlimited bandwidth, time-sensitive applications such as voice and video can

be affected by jitter, delay, and packet loss. As the transport for all session information, the

design and operation of this layer is crucial to all services, and its role is vital to the success of

the network or the user service placed upon it.

The network foundation provides an efficient, fault-tolerant transport that differentiates between

applications to allow each a fair share of the resource, yet still maintains a desired service level.

Within the architecture, wired and wireless connectivity options provide advanced prioritization

and queuing mechanisms as part of the integrated quality of service (QoS) to help ensure optimal

use of the resource.

The LAN

The core layer of the local area network (LAN) at the headquarters site is the communications

hub of the network. It aggregates client access and provides the backbone connectivity for the

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wide area network (WAN), server room, and Internet edge, making it a critical component in the

network. The LAN needs to be highly available to support mission-critical applications and real-

time media. In the past, high availability meant paying for links that were redundant and sat

unused. With Cisco SBA, all network connections are active and carry real traffic.

The following are the benefits of a Cisco SBA-designed core LAN:

Resilient for very fast failure recovery for real-time media traffic

Reduced configuration complexity with easier troubleshooting

Full use of all network links with no links sitting idle in a redundant configuration

The access layer of the LAN also provides automated services such as Power over Ethernet Plus

(PoE+), QoS marking, and VLAN assignment for IP phones in order to reduce operational

demands. The Cisco Discovery Protocol automatically recognizes endpoints and places them in

the proper VLAN without using additional address space from the data VLAN. Video endpoints

have their own set of QoS requirements, and they work in conjunction with the voice services in

order to use the bandwidth as efficiently as possible. Using Cisco Medianet technologies, the

endpoint works together with the network to signal information about its flows, allowing the

deployment of true end-to-end QoS.

The LAN design improves network speed and availability, reduces complexity, and makes the

network easier to troubleshoot and manage. This means less downtime, and fewer network

administrators are required to operate the network.

The WAN and Remote Sites

Organizations require an uninterrupted flow of information in and out of the corporate network at

the headquarters location. Cisco SBA delivers a robust WAN with the same technology used by

some of the largest networks stay to operational on an ongoing basis. A highly available WAN

helps the flow of business information proceed without interruption.

The key component in the WAN architecture is the Cisco Integrated Services Router (ISR). It

provides the following benefits to Cisco customers:

Reduces operating expense through integrated services within a single platform, such as

voice, video, and data

Protects investment with a flexible, modular design, allowing voice and video to be added

when an organization needs them

Supports all major service-provider WAN connections, public switched telephone network

(PSTN) signaling, and ISDN types

Can carry large amounts of voice and video traffic while maintaining the other core services

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Accelerates troubleshooting and enhances the assessment of the impact of each application in

your network

The primary function of the WAN router is to move data between remote sites and headquarters.

Cisco ISR Generation 2 (ISR G2) provides the platform to deliver the growing number of

services and increased performance requirements common in remote sites.

Users need seamless access, both locally and across the WAN, to network services. Call control

servers are centralized at headquarters, reducing the number of devices needed at each location.

In the event of a WAN outage, the remote site router takes over the call processing duties until

the connection can be restored. QoS prioritizes business-critical and latency-sensitive traffic so

that voice and video performance is protected and lower-priority traffic does not interfere with

critical business functions.

Network Services

Network services operate behind the scenes and allow the user services to function or improve

reliability and efficiency. In some cases, the network may become unusable without specific

services. Consider the example in this guide of the phone system. The IP phone obtained a

network address by using an automatic addressing service, such as Dynamic Host

Configuration Protocol (DHCP). The phone converted a server name, like

CUCM_Pub1.cisco.local, to a network address by using the name resolution service, Domain

Name System (DNS). The network security services helped to guarantee that the signaling and

media information was encrypted, and malicious traffic was removed or prevented from reaching

its intended target.

Within the architecture, there are many network services—including virtualization, DHCP, DNS,

various forms of security, and media resources—that are used by the call control applications

and the network-based voicemail system.

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Virtualization

Virtualization technologies can help your organization treat all IT resources as a set of shared

services that can be combined and recombined to improve efficiency and scalability.

Cisco SBA creates a foundation for virtual services. In this design, virtual LANs (VLANs) are

used to create logical, secure, and reliable segmentation between voice, video, data, wired,

wireless, and management functions on the network. The design also supports virtual servers and

storage in the server room/data center. Cisco Unified Communications Manager

(Unified CM) and Cisco Unity Connection can be installed on virtual servers and managed using

the same VMware tools as other critical business applications.

Unified communications server virtualization with Cisco Unified Computing System (UCS)

provides the following benefits:

• Consolidated workloads, raised utilization levels, and reduced operating, capital, space,

power, and cooling expenses

• Accelerated unified communications rollouts

• Ability to move workloads dynamically within a virtualization pool for greater flexibility

• Optimized performance and service levels

• Capability to scale existing applications or deploy new ones by creating more virtual

machines from an existing pool of resources

• High-availability and disaster-recovery features

Security

Security is an integral part of every network deployment. With the need to have secure and

reliable networks, protect information assets, and meet regulatory compliance requirements, an

organization needs to deploy security services that have been designed into the network rather

than added on as an afterthought. With most networks connected to the Internet and under

constant barrage from worms, viruses, and targeted attacks, organizations must be vigilant in

protecting their network infrastructure, user data, and customer information.

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Cisco SBA provides secure remote access for phones and video endpoints via a software or

hardware client. SSL VPN offers maximum flexibility, offering secure connectivity for

employees and partners back to the internal network, even from assets outside the organization’s

control. If you deploy an existing remote access solution, the Cisco SBA architecture is flexible

and can support traditional IPsec VPN clients. A hardware client that allows for an always-on

connection can support teleworkers so that home users have the same voice and video experience

that they would have in the office.

IP Network

Cisco recommends running your collaboration traffic over a private IP network rather than a

shared public network. Using an IP network allows you to expand the communication channels

beyond the traditional voice and low-quality video to include features like presence, high-

definition video, and spatial audio. If you already have an IP network in place for data, your

natural next step will be to deploy high-quality voice and video over IP. Many organizations run

voice and video systems in a mixed environment as they move from older systems to newer ones

based on the IP protocol. As you move off of older systems that use time-division multiplexing

(TDM) and ISDN solutions, you can realize significant quality improvements and cost savings.

An IP-based solution offers lower costs, easier management, remote monitoring, and control

from across the network. It also provides higher bandwidth for calls, enabling superior audio and

video quality while offering tighter integration into the corporate IT mainstream.

With an IP network based on Cisco SBA, the ongoing costs of running voice and video calls are

minimal because you are only paying for maintenance and technical support. When return on

investment (ROI) for the initial deployment is met, any additional calls are essentially free.

Because there is no incremental cost involved, employees are more likely to use the technology.

As usage goes up, returns increase, further boosting the ROI.

Cisco Medianet

The Cisco Medianet technologies include features in routers, switches, and endpoints working

together to provide capabilities such as media monitoring and media awareness. Cisco Medianet

monitoring capabilities provide increased visibility for the network operations staff. This enables

proactive management of network resources and can help the overall user experience remain

positive. Medianet media awareness helps organizations differentiate business critical

applications for service assurance, consistency, and optimal quality of user experience—end-to-

end.

The benefits of Cisco Medianet to an organization include:

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Reduced operating costs

Simplified installation and management of video endpoints

Faster troubleshooting for voice, data, and video applications

Better investment decisions to meet business objectives: ability to assess the impact of video,

voice, and data in your network

Service-level agreement (SLA) assurance and negotiation: Ability to gather key metrics for

the service provided

Ability to differentiate business-critical applications, determine the importance of a session

based on its business value, and provide endto- end QoS

Faster end-user adoption of rich-media applications through a high quality, positive user

experience

Increased confidence for network and application operators with pre-deployment assessments

Cisco Medianet includes three complementary media monitoring technologies that operate as

network services:

Performance Monitor—Allows network operators to quickly find and identify problems,

including fault location, that impact the quality of video, voice, and data. Operators can

create application class-specific threshold crossing alerts for monitoring the business critical

applications.

In Cisco SBA this feature runs in the Cisco routers, but it is also available in Cisco switches.

Mediatrace—Discovers Layer 2 and Layer 3 nodes along a flow path. Mediatrace implicitly

uses Performance Monitor to provide a dynamic hop-by-hop analysis of media flows in real

time to facilitate efficient and targeted diagnostics.

IP SLA Video Operation (VO)—Generates realistic synthetic traffic streams that are very

similar to real media traffic. It can be used in conjunction with Mediatrace to perform

capacity planning analysis and troubleshooting even before applications are deployed.

Performance Monitor and Mediatrace enable the network operations staff to quickly and cost

effectively respond to any video conferencing quality issues. These features allow the

organization to maintain a reliable and high quality service for their video conference attendees.

The IP SLA VO capabilities allow an organization to plan for future growth in size and provided

services, as well as validate deployments after fixes and updates.

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Figure 1.1 Cisco Performance Monitor in Cisco SBA Foundation with UC and video

Cisco Medianet media awareness consists of the following technologies:

• Flow Metadata—Manages and transfers application attributes to the network, allowing

appropriate policies to be applied at each hop, end-to-end

• Media Services Interface (MSI)—Resides on endpoints, and explicitly signals application

context attributes (flow metadata) to the network

• Media Services Proxy (MSP)—Uses lightweight, deep-packet inspection techniques to snoop

standard-based signaling protocols in order to produce flow metadata attributes that can then

allow appropriate policies to be applied at each hop, end-to-end Cisco SBA only utilizes MSI in

this release. The other media awareness technologies will be added in subsequent releases.

Voice and video applications are raising new requirements in terms of higher bandwidth, lower

latency, and predictable jitter. The Cisco SBA platform components are uniquely positioned to

understand the source and destination of voice and video streams, as well as the ever-changing

capacity characteristics of the connection.

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The Cisco SBA collaboration solutions are tested over the borderless network foundation

network, and they use the recommended QoS and bandwidth control settings. The conference

and scheduling resources are centralized in the data center. The endpoints and video applications,

access, WAN, and campus networks are medianet-enabled, using highly available designs and

localized services, such as Mediatrace and Performance Monitor, in the branches whenever

possible. Features such as Flow Metadata can be used with QoS in order to create more flexible

policies that reflect the organization’s business objectives. The advantage of bringing

collaboration technologies to the Cisco SBA–validated blueprint is that the initial foundation

work remains intact because the architecture was originally designed with voice and video

communication in mind.

Digital Signage Distribution Methodologies Overview

Digital Signage Overview

The past few years have seen a shift across organizations in the type of signage used to deliver

important messages to their audiences, including customers, employees, partners, and students.

Businesses are moving away from printed signs to more dynamic, flexible, and customizable

digital signs, often referred to as digital signage, electronic billboards, or e-signage. Marketing

and advertising budgets are now being directed to this new electronic medium. Financial and

retail organizations are taking advantage of digital signage to promote products and services in

their branches and stores, and to create richer, more interactive experiences for their customers.

Other industries, including government, education, healthcare, sports, entertainment, and

transportation, are also implementing digital signage as a tool to enhance customers’ and end

users’ experiences, resulting in new uses of and demands on organizations’ network

infrastructures.

One of the critical components to a successful digital signage network deployment is a thorough

understanding of your network and bandwidth availability. High-quality video files, such as

those used in digital signage, are inherently large and can cause serious network congestion

problems if not managed correctly. Proper initial evaluation and planning can save time and

money— preventing lost productivity, poor network performance, and dissatisfied users.

This white paper articulates the different distribution architectures for digital signage content and

the challenges in deploying a digital signage system. Specifically, it addresses the Internet,

leased lines, and satellite links.

Physical Distribution Media

Wide-Area Network Distribution

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Internet Cable and DSL

In today’s businesses, the Internet has become an integrated part of the larger network

architecture. Many companies install and configure VPNs over broadband to save on leased-line

telecommunications costs. This flexibility and greater cost savings represent a trade-off,

however, and can potentially compromise service levels. DSL or cable Internet service providers

(ISPs) often throttle the amount of bandwidth that the end (edge) node can consume to

accommodate more users on a single line. You must consider this limitation in advance when

streaming content to the digital signage edge node over an open Internet connection.

Prepositioning of content and securing a proper failover solution can keep the digital signage

network operational even when connectivity is lost.

Private Network Leased Line

A private leased line has been the staple of business connectivity for years. In the early

networking market, leased lines were the only method to tie the enterprise data networks

together. With private leased lines, businesses manage their own data flows. As a result, most IT

departments meet their organization’s needs by monitoring data flows and maintaining the lines

between sites exactly at the point of saturation by reducing or increasing the committed

information rate (CIR) from their service provider.

When installing a digital signage network, you should closely monitor CIR usage statistics to

ensure the flow of video content does not affect the normal traffic flow over your network.

Private leased lines can be the best option for a digital signage network for streaming content—

but also the most costly. You can achieve a good return on investment with a digital signage

network, however, by ensuring that bandwidth is used to its maximum potential.

Satellite Satellite has been used for many years as a video distribution method. Most satellite

links are multicast-enabled and therefore are a good distribution medium for video. For instance,

satellite TV is a form of the MPEG 2 video format, the most common form of digital signage

video today.

The Dish Network uplink center in Wyoming is an example of a satellite installation that uses

video encoders to digitally encode a signal into a proprietary MPEG 2 format. The signal is sent

through multicast to the satellites, which beam the signal down to antennas (known as satellite

dishes).

The signal is then decoded by an in-house decoder and converted back to analog, a format that

can be displayed on almost any television set.

In the past, IP over satellite was a limited application because of the latency in processing and

transport of data through the satellite. However, today’s satellites can transmit and receive IP

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packets with a ―round trip‖ of approximately 0.5 second. Many organizations manage satellite

links as backup circuits if a terrestrial link interruption occurs. It is common for these circuits to

be underused because they act as backups for primary terrestrial links.

Cisco now supports a module for its integrated service routers that provides the capability to

efficiently move IP traffic across satellite links. With this new innovation, you can use satellite

links as a transport medium for digital signage—representing a great way to expand the reach of

any digital signage network.

Local-Area Network Distribution

Wired

In most situations, the optimal configuration is attaching a digital media player endpoint directly

to the local-area network. Wired networks give organizations a reliable, efficient high-bandwidth

distribution medium, enabling many different streams to be run across a wired LAN with

minimal effect on the LAN itself. A wired network is the optimal solution for connecting any

Cisco Digital Media Player (refer to the section ―Cisco Digital Signage Endpoint‖) back to its

central management system.

Wireless

You can use a wireless network if a wired solution is not possible. The combination of a Cisco

wireless access point and a Cisco Digital Media Player offers the best connection if hard wiring

is not possible. Wireless also allows you to access locations where a digital sign might be

effective but is out of range of a physical cable. For example, in a retail store, digital signs are

often placed in entrances to welcome customers, but the displays cannot be wired because of

high customer traffic.

Streaming Media

Unicast and Multicast

The Cisco Digital Media Player can accept a multicast stream from any User Datagram Protocol

(UDP) audio or video stream. To accomplish this scenario, you can set the Cisco Digital Media

Manager—the Web-based central management application for all Cisco Digital Media System

products—to listen on a specified multicast address and port (refer to Figure 1.1). Then you can

configure the Cisco Digital Media Player to accept and display the stream.

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Figure 1.2 Cisco Digital Media Manager Multicast Setting

Cisco Digital Signage Endpoint

Cisco Digital Media Player

The Cisco Digital Media Player (Figure 1.2) is an integrated component of the Cisco Digital

Media System. The Cisco Digital Media Player is a dynamic, flexible, solid-state device used for

the decoding and display of digital media—including high-definition live broadcasts, on-demand

video, Flash animations, text tickers, and other Web content—on digital signage displays. This

device is small (7.5 x 5 x 1.5 in.) and weighs about 1 lb. It allows for content playback in both

full-screen mode and into regions (divisions of screen real estate) within a screen. The regions

are customizable from the Cisco Digital Media Manager interface.

The Cisco Digital Media Player has RS-232 connections for control of virtually any market-

leading digital displays. It has a built-in GUI for device and content playback management. The

Cisco Digital Media Player is built on an embedded operating system, making it highly reliable

and low maintenance.

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Figure 1.3 Cisco Digital Media Player

Local Storage Playback (Standalone)

Directly Loading Content on the Cisco Digital Media Player

Prepositioning content is a requirement for some organizations. The Cisco Digital Media Player

4305G model allows up to 2 GB of local data storage on its built-in Secure Digital (SD) card.

For this type of distribution and storage method, you can manage the transfer of files to the Cisco

Digital Media Player in two ways:

Use the Cisco Digital Media Manager to easily deliver content and playlist files by FTP to

the Cisco Digital Media Player.

Use your own FTP client to FTP files directly to the Cisco Digital Media Player and then use

the Cisco Digital Media Player’s Device Manager GUI to instruct the Cisco Digital Media

Player to play back content from local storage.

Loading Content through the Cisco Digital Media Manager

You can use the Cisco Digital Media Manager to preposition content directly to the Cisco Digital

Media Player as a failover solution. If the Cisco Digital Media Player detects a 404- or 500-level

error for the page it is loading, it automatically plays the designated failover content.

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The Capture–Transform–Share Solution

As organizations become more dependent on video for efficient communications, they are

increasingly confronted with the operational challenges associated with the planning, deploying,

and maintaining of all the components of a video infrastructure. Additionally, the complexity of

rolling out an end-to-end video architecture is further complicated because organizations have to

deal with a globally dispersed workforce that uses an ever-growing set of video-enabled

appliances, needs to consume video under different conditions, and needs to manage an ever-

growing library of video assets.

These trends can be described as time-shifting, place-shifting, and device-shifting, all of which

require a social network for distributed media. The Capture–Transform–Share solution

effectively addresses these trends by capturing all types of video, transforming video for

consumption under a variety of conditions, and sharing video across a distributed architecture

(Figure 1.4).

Figure 1.4 Capture–Transform–Share Solution

By addressing these trends with Capture–Transform–Share, organizations realize numerous

benefits, including the reduction of operational costs, effectively tapping into human talent, and

improved responsiveness of the workforce to name a few. Figure 1.5 summarizes some of the

key benefits from the perspectives of both human and business effectiveness.

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Figure 1.5 Key Benefits of Capture–Transform–Share

The transform piece in Capture–Transform–Share delivers tremendous value and allows Cisco to

differentiate our offerings. Without the transform component, the sharing of live one-way video,

two-way interactive video, and videos on demand (VoDs) is limited to a handful of combinations

when considering video sources and endpoints, as shown in Figure 1.6.

Figure 1.6 Limited Options to Capture and Share Video

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With the addition of the transform component, the solution extends itself in the following areas:

Process videos originating from a wider array of sources

Adjust resolution of video to customize for different endpoints

Transcode media to consume on different devices

Transrate media to play back even in adverse network conditions

Layer on postproduction services that are normally cost-prohibitive

Perform video analytics to auto-detect keywords and speakers, enabling more precise search

and navigability

Figure 1.7 shows how transformation extends our video solution to diverse endpoints.

Figure 1.7 Many Options to Capture and Share Video

Solution Components

This section describes the following Capture–Transform–Share solution components:

• Cisco TelePresence Content Server: Capture Live Meetings and Transform Two-Way

Interactive, One-Way Live, and VoD Media

• Cisco MXE 3500: Transform VoD and One-Way Live Streaming Applications

• Cisco Show and Share Video Sharing Application: Share One-Way Live Media and

VoDs

Cisco TelePresence Content Server: Capture Live Meetings and Transform

Two-Way Interactive, One-Way Live, and VoD Media

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The Cisco TelePresence Content Server (Content Server) is a network appliance that enables

organizations to share knowledge and enhance communication by recording their

videoconferences and multimedia presentations for live and on-demand access (Figure 8).

The Cisco TelePresence Management System (Cisco TMS) can automatically include the

Content Server in any scheduled or impromptu event. The Content Server workflow

automatically produces high-quality videos of any standards-based conference from a multipoint

control unit (MCU), Cisco TelePresence Server, or directly from a Cisco TelePresence System

endpoint, including the video participants and any secondary content, such as a presentation.

Whether it is a university lecture, a corporate training session, an executive meeting, or any other

critical event, the Content Server streamlines the process of capturing content throughout the

organization.

Figure 1.8 Cisco TelePresence Content Server

Features and Benefits

• Creates business-quality multimedia content easily from any H.323 or Session Initiation

Protocol (SIP) videoconferencing endpoint

• Supports live and on-demand streaming

• Creates content from anywhere using Cisco Expressway technology

• Manages and distributes live or recorded content to any PC and leading portable media

devices in Flash, Microsoft Windows Media, and MPEG-4 formats

• Compatible with major distribution servers and leading corporate and education Web 2.0

portals

• Streamlines the production and distribution of professional video podcasts across the

organization

• Integrates with the Cisco Show and Share media sharing application and Cisco MXE 3500

Performance Features

Up to 1080p

Support for five concurrent calls; up to two concurrent calls can be streamed live

Videoconference bandwidth up to 2 Mbps

Synchronized streaming of video and presentation in live and on-demand modes

Unicast and multicast streaming support

Internal and external storage capabilities

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Support for Microsoft Active Directory authentication through Lightweight Directory Access

Protocol (LDAP)

Call configuration and access rights management

Ability to cluster up to 10 content servers for scalable environments

Support for Structured Query Language (SQL) Server 2008 with Content Server clusters

Ability to export and import conferences from one Content Server to another

Endpoint playback

Cisco MXE 3500: Transform VoD and One-Way Live Streaming Applications

Cisco MXE 3500 is a powerful media-processing platform that helps organizations streamline

operating costs associated with live media streaming, production, and distribution by delivering a

rich set of any-to-any media processing (Figure 1.9). For live media streaming, the Cisco MXE

3500 delivers a scalable and reliable way to easily use an organization’s existing IP infrastructure

to broadcast live events. It does so by delivering live transcoding and transrating to live IP

streams and pushing the processed content out to a variety of contentdelivery- network (CDN)

devices for scalable distribution.

Figure 1.9 Cisco MXE 3500

Simplify Workflow for Video Processing

Improve communication and collaboration with the Cisco MXE 3500 to enable video

everywhere in the enterprise. The Cisco MXE 3500 extends the reach and usefulness of video for

collaboration and communications through a collection of vital media transformation services,

along with transparent integration into media-processing workflows as video files and live

streams are created, transported, and consumed over the network. This simplified workflow

opens the door to many uses such as meetings, events, training and education, organizational

communications, safety and security, and advertising, where video enables faster business

decision making, global collaboration, and scaling of expertise.

Any-to-Any Media Adaption Services

With any-to-any media adaptation on the Cisco MXE 3500, recorded and live video content is

automatically adapted from a range of incompatible media formats, resolutions, and speeds, from

standard-definition (SD) up to full high-definition (HD), so they can be viewed on demand or

live by a wide variety of playback devices and applications, such as the Cisco Show and Share

video sharing application.

Pulse Analytics for Video

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The media analytics service on the Cisco MXE 3500 offers automated keyword tagging for

spoken words and speaker recognition in the recorded video, enabling users to easily navigate to

or search for specific content or speakers in the video.

Media Postproductions

The media postproduction capabilities of the Cisco MXE 3500 offer several unique professional

studio-quality video features that can be fully automated and applied to source media files. For

example, you can append introductory videos and trailers, watermarks, and graphic overlays to

add dynamic multilayered titles, branded graphics, subtitles, captions, and animations directly on

top of the video.

Embedded in the Network

You can enable delivery of application-independent capabilities that increase adoption of

existing business investments without changing their behavior or creating an extensive IT

burden. Part of a Cisco initiative for medianet architectures, the Cisco MXE 3500 is an important

component of a pervasive video strategy. This powerful and flexible appliance integrates easily

with many Cisco and third-party multimedia products. In addition, it provides excellent return on

investment (ROI) and investment protection with software-based upgrades.

Features and Benefits

Exceptionally high-quality media transcoding and transrating for file-based and live

multimedia assets allows for any-to-any capture and playback on the network and end

devices.

The Cisco MXE 3500 provides powerful workflow automation for processing source

multimedia streams and files.

Its professional array of video and audio enhancement options includes studio-quality

editing, graphic overlays, and watermarking.

Pulse video analytics allows you to find videos based on what is spoken and who is speaking.

An easy-to-use browser-based interface for managing content makes it easy for you to

transform videos without training or administrative assistance.

The solution offers a clustering option for high scalability and transcoding redundancy.

It supports live streaming formats including Windows Media and live MPEG-2 Transport

Stream (MPEG-2 TS) so you can deliver live streams content to Cisco Digital Signs for

communications, training, events, or other applications.

Cisco Show and Share Video Sharing: Share One-Way Live Media and VoDs

Cisco Show and Share is a webcasting and video sharing application that helps organizations

create secure video communities to share ideas and expertise, optimize global video

collaboration, and personalize the connections among customers, employees, and students with

user-generated content.

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With Cisco Show and Share application you can create live and on-demand video content and

define who can watch specific content. It offers viewer collaboration tools such as commenting,

rating, and word tagging, and it provides comprehensive access reporting.

The Cisco Show and Share application fits into your organization’s existing IP network and

helps ensure that your video content is stored securely within your IT infrastructure. It supports

established video formats including Windows Media, Flash, and the MPEG-4/H.264 standard for

VoD files. The Windows Media format is supported for PC playback for live streams, and the

MPEG-4/H.264 format is supported for both PC and Macintosh for live streams. When a Cisco

MXE 3500 is available on the network, the Cisco Show and Share application allows you to have

all files that are uploaded by the Cisco Show and Share application to be automatically

transcoded to an optimal window size and bit rate using the Flash format. These files are

automatically sent from the Cisco Show and Share server to the Cisco MXE 3500, where they

are transcoded and then retrieved by the Cisco Show and Share application for editing and

publishing.

The Cisco Show and Share application is one of the many portals that the Cisco MXE 3500 and

Content Server products use to publish content and improve content search and retrieval of the

media that we capture and transform.

Cisco Digital Media System Technical Overview:

Focus on Cisco Desktop Video

The Cisco Digital Media System (DMS) is a comprehensive suite of digital signage, Enterprise

TV, and desktop video applications that allows companies to use digital media to increase sales,

enhance customer experience, and facilitate learning. Support from Cisco’s broad Partner

Ecosystem of deployment, solution development, and content creation partners helps ensure a

successful digital media implementation. This document provides a technical overview of the

Cisco Digital Media System for Cisco Desktop Video and its main components.

The Challenge

Geographically distributed organizations are increasingly striving to deliver more compelling

and effective communications to customers, employees, partners, and students to provide richer

experiences and achieve better return on investment (ROI). They look to IT groups for solutions

that address content creation, management, delivery, and access. Advances in network

infrastructure have also improved the ability of enterprise networks to support new forms of

digital media, creating the need for flexible media-management systems.

Until now, organizations have tended to adopt disparate point products that target only small

slices of the overall digital media value chain; and putting together individual components from

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multiple vendors has resulted in complex integrations, high total cost of ownership, and limited

scalability.

The challenge of supporting multiple formats, browser types, and access methods further

intensifies the dilemma facing IT groups. Most products available today limit viewer access by

employing closed systems that require special codecs and players.

This situation has created the need for an integrated solution that can address the entire digital

media value chain while also supporting industry-leading formats for live and on-demand

content publishing.

Cisco Digital Media System Completes the Lifecycle

The Cisco Digital Media System includes an integrated set of software applications that allow for

effective management of and access to live and on-demand video: desktop video (Figure 1.10).

Taking advantage of Cisco best practices from more than a decade of video and Internet

initiatives, the Cisco Digital Media System makes it simple for organizations—including banks,

retailers, corporations, schools, hospitals, and public-sector groups—to deliver high-quality,

compelling digital media to their critical audiences.

Through the deployment of the Cisco Digital Media System, organizations are better

positioned to:

Communicate effectively with targeted customers, investors, press, and analysts

Offer live and on-demand events and meetings to geographically dispersed audiences

Deliver critical information and training to employees, suppliers, and partners

Provide educational content to students

The Cisco Digital Media System solution for desktop video delivers these benefits through three

product lines—the Cisco Digital Media Manager (DMM), the Cisco Video Portal, and Cisco

Digital Media Encoders (DMEs). These advanced solutions comprise software running on high

performance Cisco media convergence server (MCS) platforms such as the Cisco MCS 7825 and

MCS 7835 Media Convergence Servers. They can support and manage nearly all standard

streaming-media formats—from Windows Media to Adobe Flash, and H.264 (AVC/MPEG4 Part

10).

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Figure 1. 10 Cisco Digital Media System Overview

Cisco Digital Media Manager

The Cisco Digital Media Manager allows content authors to publish rich digital media to the

Cisco Video Portal through a web-based management application. The Cisco Digital Media

Manager includes the following features:

● Cisco DMM Encoder Manager: You can manage encoders directly from the Cisco Digital

Media Manager Video Portal Module web interface.

● Cisco DMM Account Manager: Cisco Digital Media Manager users can have various roles

and responsibilities, thus requiring different levels of access and permissions. With the account

manager module, you can create user accounts and administer user passwords, permissions, and

profiles.

● Cisco DMM Program Manager: Content offerings, both live and on-demand, are managed in

the program manager module.

● Cisco DMM Playlist Manager: Different content offerings are easily displayed and featured

in the Featured Playlist on the Cisco Video Portal Playlist tab.

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● Cisco DMM Category Manager: You can organize content offerings into custom categories

that represent common content characteristics such as topic, subject matter or course offering,

target audience, featured executive, and business function. Through the Cisco Video Portal, you

can easily browse for content by category using the Cisco Video Portal program guide.

● Cisco DMM Interface Manager: You can design and control the Cisco Video Portal user

interface. You can easily customize elements such as the Cisco Video Portal background and

font colors, logos, ticker messages, and features for highlighting videos or communicating

messages to the end user.

● Cisco DMM Deployment Manager: All content and data additions, updates, and other

changes in the production environment must be deployed to be reflected in the Cisco Video

Portal.

● Cisco DMM Live Event Module: With this module, webcast producers can synchronize slide

graphics with streaming video and audio, and manage or administer viewer questions submitted

during a live event.

● Active Directory: Optional authentication with Microsoft Active Directory using the

Lightweight Directory Access Protocol (LDAP) gives authenticated administrators access to the

Cisco Digital Media Manager Video Portal Module and Video Portal Reports.

● Content-level viewing security: You can define by groups who can watch what video part.

● Detailed usage reporting: All video portal viewer selection activity is stored and is available

for detailed usage reporting. The report can provide details about what viewers watched what

video—and a variety of other usage reports—during a specified timeframe.

Cisco Digital Media Manager: Live Event Module

The Cisco Digital Media Manager Live Event Module add-on to the Cisco Digital Media

Manager Video Portal Module helps live-event webcast producers synchronize graphics derived

from Microsoft PowerPoint slides with live audio and video streams through the Cisco Video

Portal. The live event module also allows Cisco Video Portal viewers to submit text-based

questions at any time during a live event to the live-event producer. The producer views all

questions coming in during the event through the Cisco Digital Media Manager Question

Manager console and at any time can choose appropriate questions and either forward them to a

presenter console window or ask the presenter verbally in real time. At the end of the live event,

the producer can publish a video on demand (VoD) complete with the audio and video stream

and the synchronized graphics to the Cisco Video Portal for anytime, anywhere playback.

Cisco Digital Media Manager: Active Directory

With the Microsoft Active Directory integration using the LDAP option on the Cisco Digital

Media Manager, administrators can manage access to the Cisco Digital Media Manager Video

Portal Module, Cisco Video Portal, and Video Portal Reports. They can manage authentication

through the Cisco Digital Media Manager Video Portal Module. Cisco Digital Media Manager

offers three types of authentication:

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No authentication (users can gain access without being challenged).

Embedded authentication (users are authenticated against Cisco Digital Media Manager

built-in user database)

LDAP authentication (Cisco Digital Media Manager synchronizes username and password

with LDAP database for authentication)

Endpoint Configuration

Describe video product models

Community Resources

Cisco provides different community resources where you can engage with support

representatives or join other community members in product discussions.

Cisco product conversation and sharing site

Join other community members in discussing features, functions, licensing, integration,

architecture, challenges, and more. Share useful product resources and best practices.

On-Premises Deployments

An on-premises deployment is one in which you set up, manage, and maintain all services on

your corporate network.

Product Modes

For all deployments, the user's primary authentication is to a presence server. You must

provision users with instant messaging and presence capabilities as the base for your

deployment. You can then provision users with additional services, depending on your

requirements.

Full UC

To deploy full UC, you enable instant messaging and presence capabilities. You then provision

users with devices for audio and video in addition to voicemail and conferencing capabilities.

Cisco Jabber for Everyone (IM Only)

To deploy Cisco Jabber for everyone, you enable instant messaging and presence capabilities.

You can optionally provision users with desk phone devices that they can control with the client.

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Diagram with Cisco Unified Presence

The following diagram illustrates the architecture of an on-premises deployment that includes

Cisco Unified Presence:

Figure 2.1: On-Premises architecture

The following are the services available in an on-premises deployment:

Presence

Users can publish their availability and subscribe to other users' availability through Cisco

Unified Presence.

Instant Messaging

Users send and receive instant messages through Cisco Unified Presence.

Audio Calls

Users place audio calls through desk phone devices or on their computers through Cisco Unified

Communications Manager.

Video

Users share their screens and place video calls through Cisco Unified Communications Manager.

Voicemail

Users send and receive voice messages through Cisco Unity Connection.

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Conferencing

Integrate with one of the following:

Cisco WebEx Meeting Center

Provides hosted meeting capabilities.

Cisco WebEx Meetings Server

Provides on-premises meeting capabilities.

Diagram with Cisco Unified Communications IM and Presence

The following diagram illustrates the architecture of an on-premises deployment that includes

Cisco Unified Communications IM and Presence:

Figure 2.2: On-Premises architecture

The following are the services available in an on-premises deployment:

Presence

Users can publish their availability and subscribe to other users' availability through Cisco

Unified Communications IM and Presence.

Instant Messaging

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Users send and receive instant messages through Cisco Unified Communications IM and

Presence.

Audio Calls

Users place audio calls through desk phone devices or on their computers through Cisco Unified

Communications Manager.

Video

Users share their screens and place video calls through Cisco Unified Communications Manager.

Voicemail

Users send and receive voice messages through Cisco Unity Connection.

Conferencing

Integrate with one of the following:

Cisco WebEx Meeting Center

Provides hosted meeting capabilities.

Cisco WebEx Meetings Server

Provides on-premises meeting capabilities.

Cloud-Based Deployments

A cloud-based deployment is one in which Cisco WebEx hosts services. You manage and

monitor your cloud-based deployment with the Cisco WebEx Administration Tool.

Cloud-Based Diagram

The following diagram illustrates the architecture of a cloud-based deployment:

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Figure 2.3: Cloud-Based architecture

The following are the services available in a cloud-based deployment:

Contact Source

The Cisco WebEx Messenger service provides contact resolution.

Presence

The Cisco WebEx Messenger service lets users can publish their availability and subscribe to

other users' availability.

Instant Messaging

The Cisco WebEx Messenger service lets users send and receive instant messages.

Conferencing

Cisco WebEx Meeting Center provides hosted meeting capabilities.

Hybrid Cloud-Based Diagram

The following diagram illustrates the architecture of a hybrid cloud-based deployment:

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Figure 2.4: Hybrid cloud-based architecture

The following are the services available in a hybrid cloud-based deployment.

Contact Source

The Cisco WebEx Messenger service provides contact resolution.

Presence

The Cisco WebEx Messenger service lets users can publish their availability and subscribe to

other users' availability.

Instant Messaging

The Cisco WebEx Messenger service lets users send and receive instant messages.

Conferencing

Cisco WebEx Meeting Center provides hosted meeting capabilities.

Audio Calls

Users place audio calls through desk phone devices or on their computers through Cisco Unified

Communications Manager.

Video

Users share their screens and place video calls through Cisco Unified Communications Manager.

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Voicemail

Users send and receive voice messages through Cisco Unity Connection.

On-Premises Service Connections

Learn how Cisco Jabber for Windows can discover and connect to services in on-premises

deployments.

Connection Settings

Users set the presence server address in the Connection Settings window. Cisco Jabber for

Windows can then connect to the presence server to authenticate users and retrieve service

profiles.

Bootstrap File

You can specify the presence server address during installation with the following argument:

ADDRESS.

The installation program then saves the presence server address to a bootstrap file. Cisco Jabber

for Windows gets the presence server address from the bootstrap file when it starts. It can then

connect to the presence server to authenticate users and retrieve service profiles.

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Presence Server Discovery

Cisco Jabber for Windows can automatically discover either Cisco Unified Presence or Cisco

Unified Communications IM and Presence if you do not specify the presence server address

during installation.

When the client launches for the first time, it retrieves the presence server type from the

bootstrap file.

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• The bootstrap file contains the settings you specify during installation.

• You set the presence server type as the value of the TYPE argument during installation. In on-

premises deployments, the value must be CUP.

To discover the presence server, the client must first determine the domain. It attempts to retrieve

the domain from the following locations, in order of priority:

1. Environment variable: USERDNSDOMAIN

2. Registry key:

HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Tcpip\Parameters\Doma

in

3. Registry key:

HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Tcpip\Parameters\Dhcp

Domain After it finds the domain, the client gets the presence server address from the

Domain Name Server (DNS).

When the client gets the presence server address, it connects to the presence server and then

caches the address of the presence server.

If a redirect occurs to another server in the cluster, the client caches the address of the presence

server to which it connects, not the address of the server before the redirect.

DNS SRV Records

Cisco Jabber for Windows retrieves the _cuplogin._tcp SRV record from the Domain Name

Server (DNS) to lookup either Cisco Unified Presence or Cisco Unified Communications IM and

Presence.

• You must add this SRV record to the DNS server on the presence server domain.

• Cisco Jabber for Windows uses port 8443 to connect to Cisco Unified Presence.

• Cisco Jabber for Windows supports weight and priority in SRV records.

The following is an example SRV record:

_cuplogin._tcp.domain SRV 0 1 8443 cup_server.domain

Connect to Available Services

The client connects to available services after it retrieves the service profiles.

If the profile contains conferencing settings, the client connects to the conferencing service.

If the profile contains voicemail settings, the client connects to the voicemail service.

If the profile contains settings for Cisco Unified Communications Manager, the client does

the following:

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Retrieves the device list for the user.

Retrieves the device configuration from the TFTP server.

Registers with Cisco Unified Communications Manager.

Cisco TelePresence System 500 Series

When a customer needs immediate attention, projects arrive at a critical point, or the team is

simply dispersed, executives and team leaders can benefit from immediate access to telepresence

from their office. The Cisco TelePresence System 500 Series extends the virtual in-person

experience pioneered by Cisco’s fully immersive three-screen solutions to the executive’s or

team leader’s office. Now, resolving problems, making faster decisions, or simply checking in—

face-to-face—is just one button away.

Figure 2.5 The CTS 500 Series brings the Cisco TelePresence experience to the personal office

Extending the Immersive Experience

The Cisco TelePresence System 500 Series is ideal for joining large, multilocation group

meetings or hosting intimate, remote one-on-one discussions from the personal office. The

smaller footprint of the Cisco TelePresence System 500 gives your organization the flexibility to

easily deploy telepresence in spaces such as the personal office.

Weekly meetings, status update meetings, and calls to an expert are possible with a quick push of

a button whether scheduled or at a moment’s notice. By turning the display on its swivel mount,

the Cisco TelePresence System 500 easily accomodates another colleague joining a meeting in

the office.

Cisco TelePresence System 500 Series

Features and Benefits

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• Connect face-to-face at a moment’s notice: Combining vivid 1080p30 video and superior

audio, the Cisco TelePresence System 500 Series brings a lifelike, in-person experience to long-

distance meetings.

The integrated lighting of the system eliminates facial shadows to present a natural appearance.

• Work and share naturally: You can use the integrated single-screen (32- or 37-inch options)

as a secondary PC monitor or for video output for digital media. While not in a telepresence call,

the camera on the 32-inch screen raises automatically for full-screen real estate.

You can share content in presentation-in-picture mode or add a second screen as a dedicated

content display.

• Enjoy personalized flexibility: The easy height adjustment on pedestal and tabletop (37-inch

screen only) mount options allow you to adjust the display to suit your personal preference and

optimize virtual eye contact. The display swivels to allow an additional person to participate

within the office.

The Cisco Telepresence System 500 Series consists of the 32- and 37-inch-screen options, which

share virtual in-person features, including:

Resolutions of 1080p and 720p on a premium 32- or 37-inch screen help ensure the

telepresence call is clear and natural.

The specially designed, high-quality camera provides high-definition images,

enabling excellent eye contact among participants; the camera auto-retracts when not

in a call, leaving the full screen available for multipurpose use (on the 32-inch system

only).

Full-duplex, CD-quality audio provides a rich listening experience with no

perceivable latency or interference from mobile devices or cell phone

You have a choice of open microphone and speaker or an optional privacy headset; the 32-

inch-screen system incorporates an advanced two-dimensional microphone array for

additional audio clarity.

You can use the system when not in a telepresence call as a secondary PC monitor or for

video output for digital signage.

Integrated lighting eliminates facial shadows and provides natural-looking video.

The system is available with a pedestal, or for additional deployment flexibility, a choice of

wall-mount and tabletop options (37-inch screen system only).

Simple ―one-button-to-push‖ calling integrates with common calendaring programs.

Presentation-in-picture allows for easy content sharing; you can add an optional second

display for dedicated sharing content.

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The system offers any-to-any interoperability with standard- and high-definition video

conferencing endpoints; it also supports collaboration with desktop video applications, such

as Cisco WebEx OneTouch and the Cisco Digital Media System.

The Cisco TelePresence Total Solution

Cisco TelePresence conferencing applications empower everyone, everywhere to be more

productive through face-to-face collaboration. Cisco offers one of the industry’s broadest

portfolio of telepresence services and solutions, advancing users’ business communications and

enabling a new way of working that puts people at the center.

Describe environment recommendations

Hardware Requirements

Installed RAM

1.87 GB RAM on Microsoft Windows XP 32 bit with Service Pack 3 2 GB RAM on Microsoft

Windows 7 Free Physical Memory 128 MB Free Disk Space

256 MB CPU Speed and Type Mobile AMD Sempron Processor 3600+ 2 GHz

Intel Core2 CPU T7400 @ 2. 16 GHz GPU Directx 9 on Microsoft Windows XP 32 bit with

Service Pack 3 Directx 11 on Microsoft Windows 7

I/O Ports USB 2.0 for USB camera and audio devices.

Software Requirements

For successful deployment, ensure that client workstations meet the software requirements.

Operating Systems

You can install Cisco Jabber for Windows on the following operating systems:

• Microsoft Windows 7 32 bit

• Microsoft Windows 7 64 bit

• Microsoft Windows Vista 32 bit

• Microsoft Windows Vista 64 bit

• Microsoft Windows XP 32 bit with Service Pack 3

On-Premises Servers

Cisco Jabber for Windows supports the following on-premises servers:

• Cisco Unified Communications Manager version 7.1(4) or later

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• Cisco Unified Presence version 8.0.3 or later

• Cisco Unity Connection version 8.0 or later

• Cisco WebEx Meetings Server version 1.1 or later

Cisco Jabber for Windows supports the following features with Cisco Unified Survivable

Remote Site Telephony version 8.5:

• Basic call functionality

• Ability to hold and resume calls

High Availability for Instant Messaging and Presence

High availability refers to an environment in which multiple nodes exist in a sub-cluster to

provide failover capabilities for instant messaging and presence services. If one node in a sub-

cluster becomes unavailable, the instant messaging and presence services from that node failover

to another node in the sub-cluster. In this way, high availability ensures reliable continuity of

instant messaging and presence services for Cisco Jabber for Windows.

Cisco Jabber for Windows supports high availability with the following servers:

• Cisco Unified Presence version 8.5 and higher

• Cisco Unified Communications IM and Presence version 9.0 and higher

Configure High Availability

The following topics provide information for configuring your instant messaging and presence

service for high availability:

• Cisco Unified Presence: How To Configure High Availability Cisco Unified Presence

Deployments

• Cisco Unified Communications IM and Presence: High Availability IM and Presence

deployments configuration

Configure Re-Login Parameters

Cisco Unified Presence and Cisco Unified Communications IM and Presence lets you configure

the maximum and minimum number of seconds that Cisco Jabber for Windows waits before

attempting to re-login to the server. You specify the re-login parameters in the following fields:

• Client Re-Login Lower Limit

• Client Re-Login Upper Limit

To configure these parameters on Cisco Unified Presence, see the following topics in the

Deployment Guide for Cisco Unified Presence Release 8.6 guide:

• High Availability Client Login Profiles

• Configuring the Advanced Service Parameters for the Server Recovery Manager

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To configure these parameters on Cisco Unified Communications IM and Presence, see the

following topics in the Deployment Guide for IM and Presence Service on Cisco Unified

Communications Manager, Release 9.0(1) guide:

• High Availability client login profiles

• Configure advanced service parameters for Server Recovery Manager

Impact of Failover for Clients and Services

The following topics describe the impact of failover for clients and services:

• Cisco Unified Presence: Impact of Failover to Cisco Unified Presence Clients and Services

• Cisco Unified Communications IM and Presence: Impact of failover to IM and Presence

clients and Services

Cloud-Based Servers

Cisco Jabber for Windows supports integration with the following hosted servers:

• Cisco WebEx Messenger service

• Cisco WebEx Administration Tool, minimum supported version is 7.5

• Cisco WebEx Meeting Center, minimum supported versions are as follows:

• Version T26L with Service Pack EP 20

Version T27L with Service Pack 9

• Cisco WebEx Meetings (WebEx 11)

Directory Servers

You can use the following directory servers with Cisco Jabber for Windows:

• Active Directory for Windows Server 2003 R2

• Active Directory Domain Services for Windows Server 2008 R2

• Cisco Unified Communications Manager User Data Service

• UDS is supported on Cisco Unified Communications Manager version 8.6.2 or later.

• OpenLDAP

• Active Directory Lightweight Directory Service (AD LDS) or Active Directory Application

Mode (ADAM)

Microsoft Internet Explorer

Cisco Jabber for Windows requires Microsoft Internet Explorer 7 or later. Cisco Jabber for

Windows uses the Internet Explorer rendering engine to display HTML content.

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Known Issues with Internet Explorer

• There is a known issue with the Internet Explorer 8 rendering engine on Microsoft Windows

XP. This issue might cause unexpected behavior with Cisco Jabber for Windows.

This issue affects users on Microsoft Windows XP 32 bit with Service Pack 3 only. Users on

Microsoft Windows Vista or Microsoft Windows 7 should not encounter this issue while using

Cisco Jabber for Windows.

• In cloud-based deployments that use single sign-on (SSO), an issue exists with Internet

Explorer 9. Users with Internet Explorer 9 get security alerts when they sign in to Cisco Jabber

for Windows. To resolve this issue, add webexconnect.com to the list of websites in the

Compatibility View Settings window.

Microsoft Office

Cisco Jabber for Windows supports integration with the following software:

• Microsoft Office 2007 32 bit

• Microsoft Office 2010 32 bit

• Microsoft Office 2010 64 bit

• Microsoft Exchange 2007

• Microsoft Exchange 2010

Local Contacts in Microsoft Outlook

Cisco Jabber for Windows lets users search for and add local contacts in Microsoft Outlook.

To search for local contacts in Microsoft Outlook with the client, users must have profiles set in

Microsoft Outlook. In addition, users must do the following:

1. Select File > Options.

2. Select the Integration tab.

3. Select either None or Microsoft Outlook.

To add local Microsoft Outlook contacts to contact lists in the client, local contacts must have

email or instant message addresses in Microsoft Outlook. To communicate with local contacts in

Microsoft Outlook using the client, local contacts must have the relevant details. To send instant

messages to contacts, local contacts must have an instant message address. To call contacts in

Microsoft Outlook, local contacts must have phone numbers.

Enable Calendar Events from Microsoft Outlook

You must apply a setting in Microsoft Outlook so that calendar events display in Cisco Jabber

for Windows.

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Procedure

• Step 1 Open the email account settings in Microsoft Outlook, as in the following example:

• Select File > Account Settings.

• Select the Email tab on the Account Settings window.

• Step 2 Double-click the server name.

In most cases, the server name is Microsoft Exchange.

• Step 3 Select the Use Cached Exchange Mode checkbox.

• Step 4 Apply the setting and then restart Microsoft Outlook.

When users create calendar events in Microsoft Outlook, those events display in the

Meetings tab.

Enable Presence Integration with Microsoft Outlook

To enable integration with Microsoft Outlook, you specify SIP:user@cupdomain as the value of

the proxyAddresses attribute in Microsoft Active Directory. Users can then share availability in

Microsoft Outlook.

To modify the proxyAddresses attribute, you can:

Use an Active Directory administrative tool such as Active Directory User and Computers

The Active Directory User and Computers administrative tool allows you to edit attributes on

Microsoft Windows Server 2008 or higher.

Use the ADSchemaWizard.exe utility

The ADSchemaWizard.exe utility is available in the Cisco Jabber for Windows administration

package. This utility generates an LDIF file that modifies your directory to add the proxy

Addresses attribute to each user with the following value.

You should use the ADSchemaWizard.exe utility on servers that do not support the edit attribute

feature in the Active Directory User and Computers administrative tool, such as Microsoft

Windows Server 2003. You can use a tool such as ADSI Edit to verify the changes that you

apply with the ADSchemaWizard.exe utility.

The ADSchemaWizard.exe utility requires Microsoft .NET Framework version 3.5 or higher.

Create a script with Microsoft Windows PowerShell Refer to the appropriate Microsoft

documentation for creating a script to enable presence in Microsoft Outlook.

Enable Presence with the Active Directory User and Computers Tool

Complete the following steps to enable presence in Microsoft Outlook for individual users with

the Active Directory User and Computers administrative tool:

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Procedure

• Step 1 Start the Active Directory User and Computers administrative tool.

• You must have administrator permissions to run the Active Directory User and Computers

administrative tool.

• Step 2 Select View in the menu bar and then select the Advanced Features option from the

drop-down list.

• Step 3 Navigate to the appropriate user in the Active Directory User and Computers

administrative tool.

• Step 4 Double click the user to open the Properties dialog box.

• Step 5 Select the Attribute Editor tab.

• Step 6 Locate and select the proxyAddresses attribute in the Attributes list box.

• Step 7 Select Edit to open the Multi-valued String Editor dialog box.

• Step 8 In the Value to add text box, specify the following value: SIP:user@cupdomain.

Microsoft SharePoint

Cisco Jabber for Windows supports the following versions of Microsoft SharePoint:

• Microsoft SharePoint 2007

• Microsoft SharePoint 2010

Microsoft Office 365

Cisco Jabber for Windows supports client-side integration with Microsoft Office 365 with the

following applications:

• Microsoft Office 2007 32 bit

• Microsoft Office 2010 32 bit

• Microsoft Office 2010 64 bit

• Microsoft SharePoint 2010

Calendar Integration

You can use the following client applications for calendar integration:

• Microsoft Outlook 2007 32 bit

• Microsoft Outlook 2010 32 bit

• Microsoft Outlook 2010 64 bit

• IBM Lotus Notes 8.5.1 32 bit

• IBM Lotus Notes 8.5.2 32 bit

• Google Calendar

Virtual Environments

You can deploy Cisco Jabber for Windows in virtual environments using the following software:

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• Citrix XenDesktop 5.0

• Citrix XenDesktop 5.5

• Citrix XenApp 5.0 Feature Pack 3 Enterprise Edition for Windows Server 2008 Service Pack

2 64 bit, published desktop

• Citrix XenApp 6.0 Enterprise Edition for Windows 2008 R2 64 bit, published desktop

• Citrix XenApp 6.5 Enterprise Edition for Windows 2008 R2 64 bit, published desktop

• VMWare View Connection Manager 4.6.1.640196 with

• VMWare Agent 5.0

• VMWare Client 4.6.1

• VMWare View Connection Manager 5.1.0704644 with

• VMWare Agent 5.1.0704644

• VMWare Client 5.1.0704644

Supported Functionality

Cisco Jabber for Windows supports the following functionality in virtual environments:

• Instant messaging and presence with other Cisco Jabber clients

• Desk phone control

• Voicemail

• Presence integration with Microsoft Outlook and IBM Lotus Notes

Screen Readers

Cisco Jabber for Windows is compatible with Job Access With Speech (JAWS) screen readers.

However, the user experience with screen readers is not always consistent across the application,

depending on the version of Cisco Jabber for Windows. Users who require screen readers should

always use the most recent version to ensure the best possible user experience.

CTI Servitude

Cisco Jabber for Windows supports Computer Telephony Integration (CTI) servitude, or CTI

control of Cisco Jabber for Windows from a third party application.

Supported Codecs

Supported Audio Codecs

• g.722.1

• g.722.1 32k

• g.722.1 24k

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• g.711

• g.711 A-law

• g.711 u-law

• g.729a

Supported Video Codecs

• H.264/AVC

Network Requirements

Review network requirements such as the ports the client uses to connect to services.

ICMP Requests

Cisco Jabber for Windows sends Internet Control Message Protocol (ICMP) requests to the

TFTP server. These requests enable the client to determine if it can connect to Cisco Unified

Communications Manager. You must configure your firewall settings to allow ICMP requests

from the client. The client cannot establish a connection to Cisco Unified Communications

Manager if your firewall does not allow ICMP requests.

Room Size and System-to-User Distance Recommendations

Figure 2.6 shows the minimum and maximum recommended room sizes for a CTS-500-32 and

the range of distance that is recommended between the front of the system display and the user.

Figure 2.7 through Figure 2.9 show additional room recommendations for various room

configurations.

The CTS-500-32 allows for a wide range of distance between the system and the user. Note,

however, that the size of the user increases the closer you get to the system, and become smaller

the farther away you get from the systems.

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Figure 2.6 Room Size and Seating Distance Recommendations

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Figure 2.7 Room Size and Seating Distance Recommendations — CTS-500-32 With a Two Users and a

Freestanding Desk

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Figure 2.8 Room Size and Seating Distance Recommendations —CTS-500-32 With a Single User and

Freestanding Desk

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Figure 2.9 Room Size and Seating Distance Recommendations —CTS-500-32 Placed in a Corner

Height, Width and Weight of Pedestal-Mounted CTS-500-32

Figure 2.10 shows the dimensions of an assembled CTS-500-32 pedestal mount system.

Figure 2.10 CTS-500-32 Dimensions

Height, Width and Weight of Table Stand-Mounted CTS-500-32

The following are the dimensions of the table stand-mounted CTS-500-32:

• Height: 27 inches, +/-2 inches (69 cm +/- 5 cm)

• Base Depth: 13 inches (33 cm)

• Base Width: 24 inches (61 cm)

• Weight of desktop and stand: 30 lbs (14 kg)

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• Weight of display and codec: 47 lbs (21 kg)

• Total weight of system: 77 lbs (35 kg)

Implement desktop endpoints and surveillance cameras

Phones, Headsets, and Cameras

Review the phones, headsets, and cameras that the client supports.

CTI Supported Devices

Cisco Jabber for Windows supports the same CTI devices as Cisco Unified Communications

Manager version 8.6(1). See the CTI Supported Device Matrix in the CTI Supported Devices

topic.

Supported Headsets and Speakers

Supported Cameras

COP Files for Cisco Jabber for Windows

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In certain cases, you might need to apply COP files to Cisco Unified Communications Manager.

You can download the following COP files from the Cisco Jabber for Windows administration

package on Cisco.com:

Client-Side Availability Status

This topic describes the In a meeting (according to my calendar) checkbox on the Status tab of

the Options window.

The client supports two options for setting the 'In a meeting' availability status when events occur

in your calendar:

'In a meeting' availability status comes from Microsoft Exchange Requires

Cisco Unified Presence and Microsoft Exchange integration. Applies to on-premises

deployments.

'In a meeting' availability status comes from Cisco Jabber for Windows

Applies to on-premises and cloud-based deployments.

Availability status changes to 'In a meeting' if events occur in your calendar when:

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If you select the In a meeting (according to my calendar) checkbox, the client displays only the

'In a meeting' availability status when calendar events occur in a supported calendar client, such

as Microsoft Outlook, IBM Lotus Notes, or Google Calendar. The client does not display other

availability statuses from other calendar sources.

The following statements apply to on-premises deployments:

• You must disable Cisco Unified Presence and Microsoft Exchange integration in order for

the client to set the 'In a meeting' availability status.

• The client checks if integration between Cisco Unified Presence and Microsoft Exchange

is on or off. The client only sets the 'In a meeting' availability status if integration is off.

• The Cisco Unified Presence user options page contains the following field:

• Include Calendar information in my Presence Status

This field is equivalent to the In a meeting (according to my calendar) checkbox in the client.

Both fields update the same value in the Cisco Unified Presence database. If users set both fields

to different values, the last field the user sets takes priority. If users change the value of the

Include Calendar information in my Presence Status field while the client is running, the users

must restart the client for those changes to apply.

'Offline in a meeting' availability status refers to when the user is not logged in to the client but

an event exists in the user's calendar. Events that do not appear in the user's calendar refer to

events such as ad hoc conferencing. For example, user A creates an unscheduled Cisco WebEx

meeting. User A then invites user B to that meeting by sending the meeting URL in an instant

message.

Instant Message Encryption

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Cisco Jabber for Windows uses TLS to secure XMPP traffic over the network between the client

and server. Cisco Jabber for Windows encrypts point to point instant messages and group chats.

On-Premises Encryption

The following table summarizes the details for instant message encryption in on-premises

deployments:

Server and Client Negotiation

The following servers negotiate TLS encryption with Cisco Jabber for Windows using X.509

Public Key Infrastructure (PKI) certificates with the following:

• Cisco Unified Presence

• Cisco Unified Communications IM and Presence

After the server and client negotiate TLS encryption, both the client and server generate and

exchange session keys to encrypt instant messaging traffic.

The following table lists the PKI certificate key lengths for Cisco Unified Presence and Cisco

Unified Communications IM and Presence:

XMPP Encryption

Cisco Unified Presence and Cisco Unified Communications IM and Presence both use 256 bit

length session keys encrypted with the AES algorithm to secure instant message traffic between

Cisco Jabber for Windows and the presence server.

If you require additional security for traffic between server nodes, you can configure XMPP

security settings on Cisco Unified Presence or Cisco Unified Communications IM and Presence.

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• Cisco Unified Presence: Configuring Security on Cisco Unified Presence

• Cisco Unified Communications IM and Presence: Security configuration on IM and

Presence

Instant Message Logging

If required, you can log and archive instant messages for compliance with regulatory guidelines.

To log instant messages, you either configure an external database or integrate with a third party

compliance server. Cisco Unified Presence and Cisco Unified Communications IM and Presence

do not encrypt instant messages you log in external databases or in third party compliance

servers. You must configure your external database or third party compliance server as

appropriate to protect the instant messages you log.

See the following documents for more information about compliance:

• Cisco Unified Presence: Instant Messaging Compliance Guide

• Cisco Unified Communications IM and Presence: Instant Messaging Compliance for

IM and Presence service.

Cloud-Based Encryption

The following table summarizes the details for instant message encryption in cloud-based

deployments:

Server and Client Negotiation

The following servers negotiate TLS encryption with Cisco Jabber for Windows using X.509

Public Key

Infrastructure (PKI) certificates with the Cisco WebEx Messenger service.

After the server and client negotiate TLS encryption, both the client and server generate and

exchange session keys to encrypt instant messaging traffic.

XMPP Encryption

The Cisco WebEx Messenger service uses 128 bit length session keys encrypted with the AES

algorithm to secure instant message traffic between Cisco Jabber for Windows and the Cisco

WebEx Messenger service. You can optionally enable 256 bit client-to-client AES encryption to

secure traffic between clients.

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Instant Message Logging

The Cisco WebEx Messenger service can log instant messages, but it does not archive those

instant messages in an encrypted format. However, the Cisco WebEx Messenger service uses

stringent data center security, including SAE-16 and ISO-27001 audits, to protect the instant

messages it logs.

The Cisco WebEx Messenger service cannot log instant messages if you enable AES 256 bit

client-to-client encryption.

Client to Client Encryption

By default, instant messaging traffic between the client and the Cisco WebEx Messenger service

is secure.

You can optionally specify policies in the Cisco WebEx Administration Tool to secure instant

messaging traffic between clients.

The following policies specify client-to-client encryption of instant messages:

Support AES Encoding For IM

Sending clients encrypt instant messages with the AES 256 bit algorithm. Receiving clients

decrypt instant messages.

Support No Encoding For IM

Clients can send and receive instant messages to and from other clients that do not support

encryption.

The following table describes the different combinations you can set with these policies:

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Encryption Icons

Review the icons that the client displays to indicate encryption levels.

Lock Icon for Client to Server Encryption

In both on-premises and cloud-based deployments, Cisco Jabber for Windows displays the

following icon to indicate client to server encryption:

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Padlock Icon for Client to Client Encryption

In cloud-based deployments, Cisco Jabber for Windows displays the following icon to indicate

client to client encryption:

Local Chat History

If you enable local chat history, Cisco Jabber for Windows does not archive instant messages in

an encrypted format. In order to restrict access to chat history, Cisco Jabber for Windows saves

archives to the following directory: %USERPROFILE%\AppData\Local\Cisco\Unified

Communications\Jabber\CSF\History\uri.db

Audio and Video Performance Reference

Learn about audio and video performance for Cisco Jabber for Windows.

The following data is based on testing in a lab environment. This data is intended to provide an

idea of what you can expect in terms of bandwidth usage. The content in this topic is not

intended to be exhaustive or to reflect all media scenarios that might affect bandwidth usage.

Bit Rates for Audio

The following table describes bit rates for audio:

Bit Rates for Video

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The following table describes bit rates for video with g.711 audio:

Notes about the preceding table:

• This table does not list all possible resolutions.

• The measured bit rate is the actual bandwidth used (RTP payload + IP packet overhead).

Rates for Presentation Video

The following table describes the bit rates for presentation video:

Notes about the preceding table:

• Cisco Jabber for Windows captures at 8 fps and transmits at 2 to 8 fps.

• The values in this table do not include audio.

Maximum Negotiated Bit Rate

You specify the maximum payload bit rate in Cisco Unified Communications Manager in the

Region Configuration window. This maximum payload bit rate does not include packet

overhead, so the actual bit rate used is higher than the maximum payload bit rate you specify.

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The following table describes how Cisco Jabber for Windows allocates the maximum payload bit

rate:

Performance Expectations for Bandwidth

Cisco Jabber for Windows separates the bit rate for audio and then divides the remaining

bandwidth equally between interactive video and presentation video. The following table

provides information to help you understand what performance you should be able to achieve per

bandwidth:

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Note that VPN increases the size of the payload, which increases the bandwidth consumption.

Video Rate Adaptation

Cisco Jabber for Windows uses video rate adaptation to negotiate optimum video quality. Video

rate adaptation dynamically increases or decreases video bit rate throughput to handle real-time

variations on available IP path bandwidth.

Cisco Jabber for Windows users should expect video calls to begin at lower resolution and scale

upwards to higher resolution over a short period of time. Cisco Jabber for Windows saves history

so that subsequent video calls should begin at the optimal resolution.

Quality of Service Configuration

Cisco Jabber for Windows supports two methods for prioritizing and classifying Real-time

Transport Protocol (RTP) traffic as it traverses the network:

• Deploy with Cisco Media Services Interface

• Set DSCP values in IP headers of RTP media packets

Cisco recommends deploying with Cisco Media Services Interface (MSI). This method

effectively improves the quality of experience and reduces cost of deployment and operations.

MSI also enables the client to become network aware so it can dynamically adapt to network

conditions and integrate more tightly with the network.

Cisco Media Services Interface

Cisco Media Services Interface provides a Microsoft Windows service that works with Cisco

Prime Collaboration Manager and Cisco Medianet-enabled routers to ensure that Cisco Jabber

for Windows can send audio media and video media on your network with minimum latency or

packet loss.

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Before Cisco Jabber for Windows sends audio media or video media, it checks for Cisco Media

Services Interface.

• If the service exists on the computer, Cisco Jabber for Windows provides flow

information to Cisco Media Services Interface.

The service then signals the network so that routers classify the flow and provide priority to the

Cisco Jabber for Windows traffic.

• If the service does not exist, Cisco Jabber for Windows does not use it and sends audio

media and video media as normal.

Set DSCP Values

Set Differentiated Services Code Point (DSCP) values in RTP media packet headers to prioritize

Cisco Jabber for Windows traffic as it traverses the network.

Port Ranges on Cisco Unified Communications Manager

You define the port range that the client uses on the SIP profile in Cisco Unified

Communications Manager. The client then uses this port range to send RTP traffic across the

network.

Specify a Port Range on the SIP Profile

To specify a port range for the client to use for RTP traffic, do the following:

Procedure

• Step 1 Open the Cisco Unified CM Administration interface.

• Step 2 Select Device > Device Settings > SIP Profile.

• Step 3 Find the appropriate SIP profile or create a new SIP profile.

The SIP Profile Configuration window opens.

• Step 4 Specify the port range in the following fields:

Start Media Port

Defines the start port for media streams. This field sets the lowest port in the range.

Stop Media Port

Defines the stop port for media streams. This field sets the highest port in the range.

• Step 5 Select Apply Config and then OK.

How the Client Uses Port Ranges

Cisco Jabber for Windows equally divides the port range that you set in the SIP profile. The

client then uses the port range as follows:

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• Lower half of the port range for audio streams

• Upper half of the port range for video streams

For example, if you use a start media port of 3000 and an end media port of 4000, the client

sends media through ports as follows:

• Ports 3000 to 3501 for audio streams

• Ports 3502 to 4000 for video streams

As a result of splitting the port range for audio media and video media, the client creates

identifiable media streams. You can then classify and prioritize those media streams by setting

DSCP values in the IP packet headers.

Options for Setting DSCP Values

The following table describes the options, per operating system, for setting DSCP values:

Set DSCP Values on Cisco Unified Communications Manager

You can set DSCP values for audio media and video media on Cisco Unified Communications

Manager.

Cisco Jabber for Windows can then retrieve the DSCP values from the device configuration and

apply them directly to the IP headers of RTP media packets.

Procedure

• Step 1 Open the Cisco Unified CM Administration interface.

• Step 2 Select System > Service Parameters.

The Service Parameter Configuration window opens.

• Step 3 Select the appropriate server and then select the Cisco CallManager service.

• Step 4 Locate the Clusterwide Parameters (System - QOS) section.

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• Step 5 Specify DSCP values as appropriate and then select Save.

Microsoft Windows XP Registry Setting for DSCP Values

If you deploy Cisco Jabber for Windows on Microsoft Windows XP, you must configure a

registry setting to enable the client to set DSCP values for RTP media packets.

Set DSCP Values with Group Policy

If you deploy Cisco Jabber for Windows on Microsoft Windows Vista, or a later operating

system such as Microsoft Windows 7, you can use Microsoft Group Policy to apply DSCP

values.

Set DSCP Values on the Network

You can configure switches and routers to mark DSCP values in the IP headers of RTP media.

To set DSCP values on the network, you must identify the different streams from the client

application.

Media Streams

Because the client uses different port ranges for audio streams and video streams, you can

differentiate audio media and video media based on those port range. Using the default port

ranges in the SIP profile, you should mark media packets as follows:

• Audio media streams in ports from 16384 to 24574 as EF

• Video media streams in ports from 24575 to 32766 as AF41

Signaling Streams

You can identify signaling between the client and servers based on the various ports required for

SIP, CTI QBE, and XMPP. For example, SIP signaling between Cisco Jabber for Windows and

Cisco Unified Communications Manager occurs through port 5060.

You should mark signaling packets as AF31.

Protocol Handlers

Cisco Jabber for Windows registers the following protocol handlers with the operating system to

enable click-to-call or click-to-IM functionality from web browsers or other applications:

• XMPP:

Starts an instant message and opens a chat window in Cisco Jabber for Windows.

• IM:

Starts an instant message and opens a chat window in Cisco Jabber for Windows.

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• TEL:

Starts an audio or video call with Cisco Jabber for Windows.

Describe features and functions

Configuring Cisco Unified Communications Manager for YourCisco TelePresence System

Before you can use your system, you need to configure your system in Cisco Unified

Communications Manager (Unified CM).

You can configure your system, but your device cannot place or receive calls until you complete

the Unified CM configuration. In addition, systems appear as either a Cisco TelePresence

System 500-32 or a Cisco TelePresence System 1000 in the Cisco TelePresence Administration

GUI until you complete this configuration.

Using the First-Time Setup Wizard to Set Up your System

To use the first-time setup wizard to set up your system, complete the following steps.

If you receive errors during the setup, read the notes that accompany some of the steps for

suggested resolutions to the errors.

• Step 1 Turn the on/off switch of the system to the On position.

Figure 2.11 On/Off Switch Location

The system displays green check marks on the lower right of the display to show system

initialization and the system has initialized when the system displays six check marks. In Figure

2.12, the codec is in the process of booting up and only four of the six check marks are checked.

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If you do not see all six check marks, use the following list to attempt to troubleshoot the

problem:

• If the fourth check mark does not display, your system might not have network connectivity.

Check the network port and/or the Ethernet cable going to the network port.

• If the last check mark displayed is a red ―X,‖ there has been a compact flash error; if you

receive this error, contact Cisco Technical Support.

Figure 2.12 Booting in Progress

• Step 2 After initialization completes, make a note of the IP address that displays on the

center of the screen as shown in Figure 2.13. You use that information to log in to the Cisco

TelePresence System Administration. This IP address displays until you log in to Cisco

TelePresence System Administration or use Secure Shell (SSH) to log in to the CTS-500-32.

Figure 2.13 System IP Address

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• Step 3 Make a note of the address location.

• Step 4 Open an Internet browser on a computer that is connected to the network.

• Step 5 Enter the IP address from Step 3 in the browser address bar.

The first-time setup wizard opens and the welcome screen displays.

Figure 2.14 First-Time Setup Welcome Screen

• Step 6 Click Start to start the wizard.

The Phone Verification Screen displays.

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Figure 2.15 Phone Verification Screen

• Step 7 Click Next to continue.

The phone verification program verifies the connectivity of your system. When the program

complete successfully, the systems displays the Phone Verification Complete screen.

Figure 2.16 Phone Verification Complete Screen

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• Step 8 Click Next to continue.

The Display Verification screen displays.

Figure 2.17 Display Verification Screen

• Step 9 Verify that the image that is shown on the display is the same as the image the is

displayed on the wizard.

• Step 10 If the images match, click Next.

Figure 2.18 Eye Height Setup Screen

• Step 11 Adjust the screen so that your eyes are centered in the green box.

• Step 12 Raise or lower the CTS-500-32 so that the green rectangle that displays on the screen

is at the eye level of the CTS-500-32 user.

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Figure 2.18 Adjusting the System Height

• Step 13 Click Next.

The Camera Setup Screen displays and a red rectangle appears on the display.

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Figure 2.19 Camera Setup Screen

• Step 14 Remove the camera target from the accessory kit and place it within the borders of

the red rectangle on the screen.

• Step 15 Click Next.

Camera setup begins and a progress bar displays. When the setup completes, a screen displays

that provides an example of a flickering screen.

Figure 2.20Camera Setup Flickering Target Example screen

• Step 16 If the camera target appears to flicker on the display like the example on the

wizard, click Yes. Otherwise, click No.

If you are in country that uses a 50 Hertz (Hz) power frequency (a country other than the USA,

Canada or Mexico), there might be a noticeable flicker on the screen. If you enable the flicker

reduction feature by clicking Yes, the flicker is reduced or eliminated, but the image quality is

reduced. To eliminate the flicker at its source, you can use electronic ballast instead of a magnet

ballast for the fluorescent lights at your installation. After you change the ballast for the

fluorescent lights, you can click the Disable radio button in the 50 Hz Flicker Reduction field.

• Step 17 Click Next to continue.

The Camera Setup Complete screen displays.

If camera setup failed, it is possible that the room is too dark; in this case, add more light to the

room and click Try Again to rerun the camera setup procedure.

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Figure 2.21 Camera Setup Complete Screen

Step 18 Click Next.

Step 19 The audio verification screen displays.

Figure 2.22 Audio Verification Screen

• Step 20 Click Next to start the audio verification program.

During this test, you will hear a test start tone, white noise, and a test end tone.

When Audio Verification successfully completes, the wizard indicates displays the Audio

Verification

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Complete screen.

Figure 2.23 Audio Verification Complete Screen

• Step 21 Click Next to continue.

The VGA verification screen displays. This test verifies that your video is working for shared

presentations.

Figure 2.24 VGA Verification Screen

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• Step 22 To check that your CTS-500-32 can successfully share a video presentation,

connect the VGA-to-VGA input cable from the CTS-500-32 to an external presentation

device (such as a PC).

To skip this step, click Skip; however note that you cannot share presentations with your system.

• Step 23 Click Next to start the external presentation check.

After system verification completes, the System Verification Complete screen displays.

Figure 2.25 System Verification Complete Screen

Troubleshooting and Support

Describe troubleshooting methodologies

Troubleshooting the CTS 500-32

Revised: February 2012, OL-21845-01

You may want to periodically test system components using the hardware and software tests

available in the Cisco TelePresence System (CTS) Administration Troubleshooting window.

Before You Begin

1. Obtain your IP address in one of the following ways:

– From the CTS Cisco Unified IP phone touch the following softkeys:

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Manual > more > Info

– From the Cisco TelePresence Touch 12 tap the following:

More > Status > System Status

2. Make a note of the IP address.

3. Enter the IP address in your laptop’s browser window.

4. Click Yes to accept all security connection messages.

Testing the Display

Different light sources and the amount of light in terms of lumens or watts produces different

color temperatures. These color temperatures are sometimes expressed using terms such as cool,

warm, or daylight, but can be expressed more precisely in kelvins (K) as a numeric value.

To adjust a display:

• Step 1 Log in to the Cisco TelePresence System.

• Step 2 Choose Troubleshooting > Hardware Setup.

• Step 3 Click the Displays radio button. A test image appears.

• Step 4 Click Start in the Testing box to start the adjustment process. The Current Color

Temperature test screen appears, as shown in Figure 3.1.

Figure 3.1 Color Temperature Test Screen

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Step 5 Select the color temperature of the lighting in the meeting room from the drop-down

menu. The Apply button is activated.

Step 6 Click Apply.

Step 7 Click Stop to stop the test.

Verifying Speaker Function

To verify that the speakers are working properly and that the left and right speakers are not

switched:

Step 1 Log in to the Cisco TelePresence System.

Step 2 Navigate to Troubleshooting > Hardware Setup > Speakers.

Step 3 Click Start to begin the speaker test. The speaker test screen appears, as shown in

Figure 3.2

Figure 3.-2 Speaker Test Screen

Step 4 Click Cycle through Speakers to have sound cycled automatically for 5 seconds on

each speaker.

Step 5 Listen carefully as the sound moves from the left to the right speaker and watch the

speaker icons on the display.

Step 6 Make sure the sound from the speakers corresponds with the left and right speaker

icons on the display.

Step 7 Click Stop to end testing.

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Verifying Microphone Function

Tip During a call, the CTS 500-32 microphone can transmit ambient noise in the surrounding

area, such as other people talking in or near the same area. Make every effort to reduce or

eliminate background noise before you begin the test call.

To verify that the CTS 500-32 microphone array is working properly:

• Step 1 Log in to the Cisco TelePresence System.

• Step 2 Navigate to Troubleshooting > Hardware Setup > Microphones.

• Step 3 Click Start to begin the test.

• Step 4 Lightly tap the microphone on the underside of the display and watch the audio meter

on the display to make sure that the sound registers.

• Step 5 Click Stop to end the test.

Testing the Light Fixture

Use the following information to check the LEDs, light fixture, and configure the audio for the

digital media player (DMP).

Light

Each CTS 500-32 has a built-in light fixture. This troubleshooting feature lets you see or change

the status of the light.

• Step 1 Navigate to Troubleshooting > Hardware Setup > Other Devices.

• Step 2 Click the Light tab.

• Step 3 Click the Start radio button to begin testing the light. If the light feature is enabled, the

Light State On button is highlighted. If the light feature is disabled, the Light State Off button

is highlighted. Click the Light State On or Light State Off button to change the current state

of the light.

• Step 4 To end the test, click Stop.

When you end the testing, the state of the light reverts to its default setting as specified in Cisco

Unified Communications Manager.

Testing Digital Media Player Audio

When you are not in a Cisco TelePresence call, the DMP feature allows you to use the DMP

audio as the secondary audio input source.

To configure and test the DMP audio:

• Step 1 Connect the DMP to the codec and display as follows:

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Connect the DMP HD video cable to the HD Video input marked as ―Auxiliary Input (ex.

DMP)‖ on the codec.

Connect the DMP audio cable to the Auxiliary PC Audio input on the codec.

• Step 2 Navigate to Troubleshooting > Hardware Setup > Other Devices.

• Step 3 Click the DMP tab.

• Step 4 Click the Start button.

• Step 5 Select one of the following radio buttons:

• Select the DMP radio button if the audio that you connected to the codec is coming

from a DMP.

• If you are using a non-DMP source, select the PC radio button.

• When Secondary Audio Input Source is set to DMP, audio input is active only when:

• The CTS 500-32 is not in a call

• The time period is during normal business hours

When Secondary Audio Input Source is set to PC, the audio input is active while the presentation

source is active, both in and out of a call.

• Step 6 To end DMP audio configuration or testing, click Stop.

The CTS 500-32 retains the setting that you just set.

Identify endpoint issues

Troubleshooting CTS Calls

Use the information in Table 3.1 to troubleshoot CTS call issues

Table 3.1 Troubleshooting CTS Calls

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Collect system information

Introduction

This document provides guidelines for the collection of logs and other diagnostic information to

assist in the resolution of issues with the Cisco TelePresence Video Communication Server

(Cisco VCS).

It is intended for use by the Cisco VCS system administrator or other support engineer.

Alarms

New, unacknowledged alarms are indicated in the top right corner of every VCS web page.

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To see the details of each alarm, click on the alarm indicator, or go to Status > Alarms.

The Alarms page shows the type of alarm and which peer in a cluster (if applicable) it is

affecting. It also indicates the remedial action to take to resolve the alarm. Alarms that are not

important in an installation’s particular circumstances can be acknowledged.

Alarms are also listed when logging in to the command line interface (CLI).

VCS logs

There are three types of VCS logs which can be seen by going to Status > Logs > [type]. These

are passive logs, which the administrator can view and filter, but cannot interact with in other

ways.

Event Log

The Event Log shows key events that have occurred on the VCS including call events, login

events and alarms. Red events indicate events that have failed; green indicates events that have

succeeded.

You can use the Filter options to search for specific URIs or keywords. The Event Log is the

same as the messages files in the system snapshot.

Syslog

The Event Log can also be sent to one or more external syslog servers, for remote system

monitoring. This is configured on the Logging page (System > Logging).

Up to four syslog servers can be specified.

Configuration Log

The Configuration Log provides a list of changes made to the VCS configuration by the system

and through the web interface or CLI. It also shows from which IP address and user the changes

were made.

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This log is useful when reviewing a system which has started to behave unexpectedly - any

changes made to the system can be reviewed to see if they may have had an impact on the state

of the system.

Network Log

The Network Logs are similar to the Event Logs, in that they both show SIP and H.323

messaging. However the Network Logs also shows call routing decisions made based on the

VCS search rules.

Call and search history

Calls Current call status and historical calls can be seen on the Call status and Call history pages

(Status > Calls > Calls and Status > Calls > History respectively).

• Current calls: the information shown includes the routing, bandwidth allocation and

protocol being used.

• Historic calls: release cause information is also shown.

Search history The Search history page (Status > Search history) shows the decisions the VCS made to route

a call, based on transforms, FindMe profile and search rules, zones and soon.

This information is useful if calls are not hitting their intended destinations. It assists in working

out why a call may be heading in a different direction to that which was expected.

Advanced logging levels

If instructed by the support organization, you can more finely tune the log levels before starting

diagnostic logging. This is configured on the Network log level and Support log level pages

(Maintenance > Diagnostics > Advanced > Network log configuration and Maintenance >

Diagnostics > Advanced > Support log configuration respectively).

Setting any of these log levels higher than their default will raise an alarm to indicate that the

VCS is running with a higher log level than normal. These log levels are not reset after stopping

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the diagnostic log and so must be manually reset to their default level of Info after logging is

complete.

Wireshark

You can take a TCPdump on the VCS which can then be copied off the system and analyzed in

Wireshark or similar tools.

A packet capture of all the network traffic being received and sent via the VCS Ethernet

interfaces can be saved to the VCS hard drive. The packet capture will include all network traffic

(including RTP – if the media is routed via the VCS) seen by the VCS Ethernet interface.

Note: if TLS connections are used for SIP signaling, Wireshark will only show the TLS packets,

it will not be able to decode the SIP traffic.

On VCS, log in as root and type:

mkdir /mnt/harddisk/traces

cd /mnt/harddisk/traces

Then to activate the trace type:

tcpdump -w trace.cap -s 0 -C 10

• -w instructs tcpdump to write the raw packets to file rather than parsing and printing them

out. The raw packets are (initially) written to the specified file name (in this case trace.cap).

• -s sets snaplen to 0 (which instructs tcpdump to capture complete packets regardless of

packet length).

• -C restricts the output file size to the number (following the option) in millions of bytes.

In the example above, after the initial output file has reached 10 million bytes in length (~10

MB) then a new output file is created and used. The file name will have an incremental index

appended to it (trace, trace2, trace3 and so on)

By default the tcpdump command (without the -i option specified) will collect packet data from

the lowest available interface ID, that is eth0.

To stop the packet collection press: Ctrl+C

The capture files will be available in the following directory:

/mnt/harddisk/traces/

Use an application which can do SCP to copy them to a local machine (PC). For example,

Winscp is a free SCP client for Windows.

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If after the packet capture has been stopped, the OS reports that packets have been dropped

during the capture (which could happen on very busy systems), make a note of it and let the

support organization know, if the packet trace is to be sent on to them.

Manage configuration

Troubleshooting Cisco Unified CM

Use the information in Table 3.2 to troubleshoot Cisco Unified CM issues.

Problem Possible Cause or

Description

Action

Video resolution is reduced. Codec negotiates down speed

to a lower resolution, possibly

due to one of the following:

• A lower resolution

endpoint joins the

meeting.

• A satellite connection or

extended reach endpoint

joins the meeting.

• There is not enough

bandwidth configured in

the Cisco Unified CM

Region settings.

1. Check to see if a lower

resolution endpoint or

satellite connection

endpoint has joined the

meeting.

2. Check your bandwidth

settings in Cisco Unified

CM by doing the following:

a. Log onto Cisco Unified

CM administration page.

b. Navigate to your Region

configuration to see if

enough bandwidth has

been given and assigned to

the codec.

Minimum is 16000; the Cisco

recommended value is 32000.

Multipoint calls are dropped. While in a multipoint meeting,

a CTS endpoint places a call

on hold or an audio add-in

joins the meeting. As a result,

a call is dropped.

1. Check the Cisco Unified

CM configuration to

determine if duplex

streaming mode is enabled.

2. If this mode is enabled,

disable it. Enabling this

mode causes issues with

certain CTMS

functionality, such as audio

add-in and calls on hold.

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CTS 500 shroud light does not

turn on automatically.

In some cases, the CTS 500

lights do not turn on

automatically when the Lights

Field is set to ―On All the

Time‖ in the Cisco Unified

CM Administration interface.

The lights will turn on and off

with the work time hours that

were set in the ―Display On

Time‖ field. If you want the

lights to be on all the time, be

sure that you enter the

following:

1. Light field is set to ―On All

the Time.‖

2. Display On Duration field

is set to 23:59.

Document camera cannot be

controlled by the CTS or the

phone.

Incorrect configuration

settings on the document

camera or on Cisco Unified

CM.

1. Verify that all cables are

seated properly and that

the Ethernet connection is

on.

2. Verify that the power is on

and the LED is on and is

showing green.

3. Use the remote control to

ensure that the

configuration is correct:

a. Resolution is set to

XGA/60

b. Preset Control should

be set to default

c. Protect mode should be

off.

4. Verify that the presentation

input device is enabled in

Cisco Unified CM.

5. If the Cisco Unified IP

Phone cannot control the

document camera, ping the

document camera from the

admin CLI.

6. Re-seat the Ethernet cable

to document camera.

7. If needed, use the remote

control to rule out hardware

issues. Connect the document

camera directly to the

projector to rule out projector

issues.

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CTS does not upgrade. • CTS cannot find or

download the upgrade file

from the Cisco Unified

CM TFTP server.

• AutoUpgrade is set to

false.

a. Check that the correct

upgrade file name is

configured on the CTS

Device page in Cisco

Unified CM.

b. Check whether the upgrade

file is uploaded to the

TFTP server.

c. Check whether TFTP

service has been restarted

after the upgrade file is

uploaded.

d. Check whether the CTS is

pointed to correct the

TFTP server where the

upgrade file is located.

e. Set AutoUpgrade to True.

Determine your settings by

entering the following CLI

command:

show upgrade det

If AutoUpgrade is set to False,

re-set it to True.

Contact TAC for assistance.

Cisco Unified IP Phone does

not register but CTS registers

fine.

Phone profile or Directory

Number (DN) not provisioned

properly in Cisco Unified CM.

Log onto Cisco Unified CM

and make sure that the phone

profile and Directory Number

(DN) are created and

configured properly.

Non-MIDlets Phone user

interface (UI) does not show

Cisco TelePresence-specific

information, or does not

refresh until a service button

is pushed.

Service or Authentication

URL is not configured

properly on the Cisco Unified

CM phone device page.

Log onto the Cisco Unified

CM administration phone

device page and check the

following:

1. Service on the device.

2. Authentication and Idle

URL—Verify that this

address points to the correct

codec IP address, and follow

the recommended format.

The phone does not display

the Cisco TelePresence idle

screen.

• Phone may not be

• recognized:

1. Check the cable

connection from the

primary unit to the Cisco

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• Cisco Unified CM

does not know about it.

• CTS is not registered

because it is

unplugged.

• The phone did not

receive an IP address.

• There could be errors in

the

• Cisco Unified CM Phone

• Configuration window:

• Incorrect IP address.

• Typos in the external

location URLs.

Unified IP Phone.

2. Verify phone registration

by logging in to the Cisco

Unified CM administration

page.

a. Click on the IP address

and verify phone

registration.

b. Verify the phone in the

system.

3. Correct typos in the URL.

Cisco Unified IP Phone does

not register.

• Cisco Unified CM does

not know about it.

• CTS is not registered

because it is unplugged.

• CTS MAC address is

entered incorrectly.

• The incorrect device type

was configured in Cisco

Unified CM.

1. Verify phone registration

by logging into the Cisco

Unified CM administration

page.

2. Click on the IP address and

verify phone registration.

3. Confirm that a Cisco

Unified IP Phone 7970 or

7975 device type has been

configured in Cisco

Unified CM.

• MIDlets does not start up.

• Phone screen still shows

regular IP phone UI

instead of expected CTS

phone UI.

• Incorrect Phone Service

Name.

• Incorrect Service URL in

Phone Service.

• Incorrect Service Vendor in

Phone Service.

1. Create Phone Service

with the correct name:

a. Unsubscribe the

phone from the old

service.

b. Subscribe the

phone to the new

service.

2. Update the Phone

Service based on

recommended format,

run Update

Subscriptions.

3. Update the Service

Vendor to ―Cisco,‖

then run Update

Subscriptions.

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MIDlets fails to start up

properly, shows the following

message:

―setting up network

connections‖

MIDlet cannot establish

TCP connections with the

CTS.

Possible incorrect CTS IP

address is assigned to the

phone profile

authentication server URL.

1. Ensure that the correct

CTS IP address is in the

2. Authentication Server

URL for the phone device

in Cisco Unified CM.

Click Save.

3. Reset the phone.

MIDlets fails to initialize

properly, showing the

following message:

―Configuration error.‖

• The MIDlet does not find

Authentication Server

URL or it does not

recognize its format.

• Non-English characters

are used on phone device

profile.

1. Properly configure the

Authentication Server

URL for the phone.

2. Click Save.

3. Reset the phone.

CTS was moved to a different

Cisco Unified CM and the

registration is rejected.

The CTS was associated with

a different secure Cisco

Unified CM at one time and

the CTS preserved the

previous Certificate Trust List

(CTL) file.

Remove the CTL file from

admin graphical user interface

(GUI) in the Cisco Unified

CM Administration interface.

The Cisco TelePresence unit

does not register.

Cisco TelePresence System

could be unknown:

• Cisco Unified CM does

not know about the CTS.

• CTS is not registered

because it is unplugged.

• CTS MAC address is

entered incorrectly.

1. Test the network

connection to the master

codec by doing the

following:

a. Plug the codec network

cable directly into the IP

phone.

b. If the IP address displays,

the problem is with the

codec.

2. Verify the phone

registration by doing the

following:

a. Log in to the Cisco Unified

CM Administration

interface.

b. Click on the IP address and

verify the phone

registration.

CTS does not register with

Cisco Unified CM:

• From the Cisco Unified

CM Device page, the CTS

• CTS profile is not

provisioned properly in

Cisco Unified CM.

• Directory Number (DN)

is not configured.

• Log onto Cisco Unified

CM, make sure that the

CTS profile and the

directory number (DN) are

created and configured

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status shows unregistered

or unknown.

• From the CTS codec Web

user interface (UI), CTS

status shows unknown or

inaccessible for Cisco

Unified CM.

properly.

• Completely delete from

Cisco Unified CM the

CTS and the phone,

including its associated

DN, then add them back to

Cisco Unified CM.

• Cisco Unified CM or

TFTP service issue.

• TFTP port 6970 is blocked

so that the CTS cannot

download the ―device

Config xml‖ file from

Cisco Unified CM TFTP

server.

• XML configuration file is

suspected to be corrupted

on the Cisco Unified CM

database. Cannot resolve

hostname of Cisco Unified

CM.

• Make sure Cisco Unified

CM and TFTP service is

running. Restart services if

necessary.

• Make sure there is no

firewall or device between

the CTS and Cisco Unified

CM that blocks the 6970

port.

• If you are using the Cisco

Unified CM hostname as

the TFTP server on the

CTS, make sure that the

hostname can be resolved

by the domain name

system (DNS).

CTS un-registers from time to

time.

• The system experiences a

SIP registration timeout.

• Intermittent network issues

could cause packets to be

dropped.

1. Confirm that Cisco Unified

CM is receiving SIP

messages and whether the

system is responding.

Collect a packet capture if

necessary to submit to

2. Cisco technical response

for further review.

• Main display: Adjusting to

lower video resolution to

match available resources.

• Call ended due to video

quality configuration

mismatch.

• Video quality has been

downgraded to match the

available

resource/configuration. For

example, if the 1080p

endpoint calls a 720p

endpoint, the bandwidth of

the 1080p endpoint is

downgraded to 720p.

• The call is ended due to a

video quality configuration

mismatch. For example,

the 720p endpoint tries to

join the Cisco

Verify that the device or

resource is configured

properly in Cisco Unified CM.

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TelePresence Multipoint

Switch (CTMS) 1080p

conference.

Time does not show correctly

on the CTS or phone.

Network Time Protocol (NTP)

is not configured properly or

the codec does not sync up

with NTP.

1. If NTP is not configured,

access Cisco Unified CM

date/time group, configure

NTP properly and assign to

a CTS device pool.

a. 2. Make sure that the CTS can

ping NTP, and there is no

firewall blocking the 123 NTP

port.

SNMP query does not work. SNMP is not configured

properly in Cisco Unified CM

or the configuration is not

accepted by the CTS.

1. Make sure SNMP is

configured properly in

Cisco Unified CM and that

the configuration has been

pushed to the CTS.

2. Modify the SNMP

configuration in Cisco

Unified CM so that the

updated configuration can

be pushed to the CTS.

3. Reboot the CTS if

necessary.

Implement key CLI commands

Troubleshooting CTS System Administration

Use the information in Table 3.3 to troubleshoot system issues, including the administration Web

interface.

Table 3.3 Troubleshooting CTS System Issues

Problem Possible Cause or

Description

Action

• Cannot ping primary

codec.

• Phone shows ―requesting

service‖ message.

• Display shows the IP

address 192.168.100.2

when codec boots up.

There is no DHCP Server

available on the network.

The codec looks for DHCP by

default during first-time

bootup. If there is no DHCP

available, the codec times out

or gets stuck in the boot up

1. Connect your laptop to the

camera port on the codec.

The laptop should get a

DHCP IP address

assigned.

2. Use SSH to access the

codec and restart calling

b. service by entering the

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sequence. following admin CLI

c. command:

d. 3. Once service is up and

running, log in to the CTS

Web UI and configure the

CTS to use static IP rather

than DHCP.

• Cannot ping the primary

codec or the phone from

the switch port.

• On the switch, Cisco

Discovery Protocol (CDP)

data is received about the

codec and the phone.

The switch port is configured

for the same VLAN ID on

both voice and access VLAN,

which is not recommended.

On the switch to which the

codec is connected, configure

a different VLAN ID for the

following:

• Voice

• Access VLAN

• Unable to reach the

secondary codec.

• Web UI shows a red X

on the secondary

codec.

If the primary codec cannot

ping the secondary codec:

• The secondary codec may

be down.

• There could be another

hardware failure.

If the primary codec can ping

the secondary codec, but the

primary codec still indicates

that the secondary codec is not

reachable, there may be

nonvolatile (NV) corruption or

another software related issue.

1. Check the physical cable

connections.

2. On multi-screen systems,

reboot all three codecs and

verify that the correct IP

address shows on all three

main display screens upon

reboot.

3. If you are denied SSH

access to the codec

through the camera port

using your administration

account, the problem may

be NV corruption. Contact

TAC for assistance.

4. If SSH access is granted,

try a factory reset on the

secondary codec.

Upon CTS bootup, the system

does not pass all six green

checks, or gets stuck on one of

them.

This may be hardware failure

or non-volatile (NV)

corruption.

Contact TAC for assistance.

CTS reboots by itself without

user intervention.

• Unstable power source.

• Kernel panic: Check

whether kernelpanic.log

exists in the /nv/log

directory.

1. Check your power source

and cabling.

2. If a kernelpanic.log exists:

a. Analyze the log for the

possible root cause.

b. Check and re-seat all

Ethernet cable

connections; networks

experiencing route

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flapping might consume

significant amounts of

memory resources leading

to kernel panic.

3. Check and re-seat the

Ethernet connection

between the primary and

secondary codecs. Reboot

the primary codec if

necessary.

• CTS locks up and access

through SSH or GUI gives

error.

• While the system responds

to regular ping requests, an

―internal server error‖

shown on the Web UI.

Possible memory leak or

system resource exhausted:

Some process consumes

majority of memory resources

or hog CPU resource, hence

system locks up.

1. If you can still access the

system through SSH,

enter the following CLI

command to get a snapshot

of the system state:

show tech runtime

2. Reboot the codec if

necessary to recover the

system.

The Web GUI denies

username/password login, but

SSH access is accepted.

Login issues:

• The codec is in the process

of booting up or upgrading.

• Calling service is not

running.

• There is password file

corruption on the GUI

access.

1. Wait until boot or upgrade

is complete.

2. Restart calling service using

the admin CLI.

See the Cisco TelePresence

Administration

Software Command

References home page on

Cisco.com for information

about CLI commands.

3. Factory reset the codec only

as a last resort.

Contact TAC for assistance if

necessary.

Connectivity to codec keeps

dropping, users are kicked out

of SSH or Web GUI session

periodically.

Possible duplicate IP address

on the network that kicks

users out of codec

connectivity.

1. Go to another computer

and do the following:

a. Disconnect the uplink

Ethernet cable on the

CTS.

b. Ping the IP address. If this

ping is successful, there is

a duplicate IP address.

a. 2. Identify the device with the

duplicate IP address to correct

the issue.

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Time does not show correctly

on the CTS or phone.

Network Time Protocol (NTP)

is not configured properly or

the codec does not sync up

with NTP.

1. If NTP is not configured,

access Cisco Unified CM

date/time group, configure

NTP properly and assign

to a CTS device pool.

2. Make sure that the CTS

can ping NTP, and there is

no firewall blocking the

123 NTP port.

SNMP query does not work. SNMP is not configured

properly in Cisco Unified CM

or the configuration is not

accepted by the CTS.

1. Make sure SNMP is

configured properly in

Cisco Unified CM and that

the configuration has been

pushed to the CTS.

2. Modify the SNMP

configuration in Cisco

Unified CM so that the

updated configuration can

be pushed to the CTS.

3. Reboot the CTS if

necessary.

Cannot download logs from

Web UI.

Not enough disk space. 1. Make sure disk space is

not full.

2. Check disk space using the

following CLI command:

show status

Use the following admin

CLI commands to check

3. whether any previous logs

have been generated:

utils logs status

Use the following admin

CLI command to see if

4. logs can be sent out using

FTP:

utils logs ftp

Peripheral shows red X on

Web GUI, sometimes

triggering SNMP alarms.

Could be introduced by

transient state changes on

various CTS components or

false alarms.

Reboot the codec to clear up

the error condition.

Status Details

Window—Microphone and

Speaker status show false

positives (green check-mark,

when they should show red

Administration Web interface

issues:

There is no way to determine

what the status of the

To determine individual

microphone and speaker

functionality, use the hardware

troubleshooting interface for

your system.

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X). microphones that are

connected directly to the

codec, only the ones

connected to the Audio

Extension Box.

Resetting the CTS Factory Image

When you change the CTS model while performing configuration tasks in the administration

interface (switching from a CTS 1000 to a CTS 1300-65, for instance), the CTS must be

rebooted.

Monitor events and alerts

No-Show Meetings and Meetings without Usage Survey Responses

This email is sent if there are any meetings that were scheduled but never took place (no-show

meetings), and/or meetings for which the survey was not completed by the meeting organizer.

The following information is provided:

Table 3.4 Organizers of No-Show Meetings

Table 3.5 Meetings without Usage Survey Responses

Mailbox Alert

This email is sent if the CTS-Manager mailbox has exceeded it size limit and is no longer able to

send emails to meeting organizers. In this situation, it is important to delete all emails no longer

required. Cisco recommends setting policies to automatically purge emails when the mailbox

reaches 75% of its allocated threshold.

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Table 3.6 contains mailbox quota information.

Certificate Expiry

This email is sent if one or more security certificates are about to expire. In this situation, it is

important to replace the expiring certificate(s) as soon as possible, so CTS-Manager will

continue to function properly.

Table 3.7 contains certificate expiry information.

Table 3.7 Certificate Expiry Information

Conferencing Concepts

Describe multi point control units

MCU overview

An MCU is predominantly used to connect SIP or H.323 based single-screen endpoints into

virtual meeting rooms.

The number of ports on the MCU limits the total number of concurrent participants. The number

of ports is dependent on the model of MCU/number of blades in the Cisco MSE 8000, the

licenses they have applied to them and the mode in which they are running.

Conference initiation

Conferences can be initiated on an MCU in a number of ways detailed below; however, not all of

them are available in every deployment.

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A resource used for scheduled conferences should not be also used for ad hoc conferences in

order to guarantee port availability for scheduled calls. Therefore Cisco recommends that MCUs

used for scheduled conferences are never used for ad hoc calls and separate MCUs are provided

for ad hoc conferencing.

Scheduled conferences

Scheduled conferences are pre-booked conferences with a start and end time and a pre-defined

set of participants. MCU scheduled conferences are booked via TMS, either using TMS directly

or via integration point such as Microsoft Exchange.

Non-scheduled conferences

There are various means of creating or joining an ad hoc MCU conference. These methods are

not supported on MCUs that TMS uses for scheduled calls, and some methods are only

supported when the MCU is deployed in a certain way, as detailed below.

The MCU auto attendant

The MCU auto attendant is an interactive menu that is displayed when users dial the MCU’s auto

attendant number. It can be used to create a new conference or to join one of the existing ones.

More than one auto attendant can be configured, each with a unique dial-in number.

The auto attendant is not supported when the MCU is deployed on Unified CM or when using

Conductor.

Dynamic escalation conferences

There are two mechanisms that support the ability to escalate from a point-to-point call to a

multipoint call hosted on an MCU. Multiway is the VCS based mechanism that can only be

initiated by endpoints that support Multiway. Unified CM also supports a mechanism that

requires the endpoint to support the conference button in order to escalate the call.

Multiway escalation is only supported when using an ad hoc MCU registered to VCS. Conductor

does not fully support all Multiway features. Similarly, escalation using the Unified CM method

must be to an MCU configured as a media resource on Unified CM.

Rendezvous conferences

Rendezvous conferences on an MCU are those that a participant can join at any time. These conferences

can be configured for individual use, or for communal first-come, first-served conferences.

Rendezvous conferences can be statically configured on an MCU by defining a conference room on the

device. It is also possible to dynamically create a conference room so that no pre-configuration is

required. Statically configured conferences allow unique settings to be set per conference, whilst dynamic

conferences must follow a single template.

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When using the MCU with VCS but not Conductor, static conferences must be defined on individual

MCUs and therefore are vulnerable to a single point of failure.

When using Conductor, Rendezvous conferences are configured on the Conductor; therefore the

conference is never statically defined on a single MCU. This increases conference resilience while

maintaining the ability to have unique conference settings.

When an MCU is registered to Unified CM, it is also possible to make Rendezvous conferences. The

administrator defines a range of numbers that can be used for Rendezvous conferences, and when users

require a conference they press a ―Meet Me‖ button and choose a conference number to start a

conference.

Network topology

An MCU causes a concentration of video traffic at its location because each port can have a video call

connected to it at up to 4Mbit/s (plus 20% overhead). Therefore, MCUs should be placed at a network

location that has enough bandwidth to host these calls.

Cisco recommends that MCUs be placed on the internal network with firewall protection from outside

access. For external calling, a Cisco TelePresence VCS Expressway should be used in conjunction with a

VCS Control in order to allow video calls to traverse the firewall.

If the second Ethernet port is activated (on the MCU 4500 and 5300 series this requires the Video

Firewall Option key), Cisco recommends that this port is also on the internal network and used for

purposes such as separating MCU management traffic from MCU video traffic.

Describe conferencing features

Deploying an MCU as a Unified CM media resource

Deployment overview

This deployment uses the media resource management capabilities of Unified CM in order to provide ad

hoc calling capabilities. Calls can either be dynamically escalated using the conference button or

Rendezvous based using the ―Meet Me‖ button.

Step-by-step configuration is available in the Unified CM Administrator and System guides, listed below.

However an overview of the deployment process follows.

Configuration Steps

Step 1: On the MCU, configuring settings

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1. Go to Settings > Conferences.

2. Configure the fields on the MCU as follows:

Table 4.1 MCU settings when registered to Unified CM

3. Click Apply changes.

Step 2: On Unified CM, configuring conference features

To configure conference features such as the maximum number of participants:

1. Go to System > Service Parameters.

2. Select the relevant Unified CM Server.

3. Select the Cisco CallManager (Active) as the service.

4. Select Advanced to show advanced options.

5. Configure the Clusterwide Parameters (Feature - Conference) section as required.

Step 3: On Unified CM, adding the MCU

Add the MCU to Unified CM as a manageable device as follows:

1. Go to Media Resources > Conference Bridge.

2. Click Add New.

3. Select Conference Bridge Type as Cisco TelePresence MCU.

4. Enter relevant fields and click Save.

Step 4: On Unified CM, configuring a media resource group list

1. Go Media Resources > Media Resource Group.

2. Click Add New.

3. Choose a name and move the MCU(s) into the Selected Media Resources area.

4. Click Save.

5. Go to Media Resources > Media Resource Group List.

6. Click Add New.

7. Choose a name and move the created Media Resource Groups into the Selected Media Resource

Groups area.

8. Click Save.

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Step 5: On Unified CM, assigning a Media Resource Group List to a device

1. Go to Device > Phone.

2. Select a device.

3. Choose the Media Resource Group List that you created earlier.

4. Click Save.

Describe scheduling vs ad hoc vs on-demand features

Deploying an MCU registered to Cisco TelePresence VCS

Deployment overview

This deployment uses VCS as the registration mechanism for the MCU and conferences are

scheduled using TMS: separate MCUs are used for scheduling and ad hoc conferencing.

Endpoints registered to VCS (or a Unified CM trunked to the VCS) can join MCU calls. All

MCUs are provided with a unique prefix in order to route calls correctly and avoid the possibility

of identical conference numbers for scheduled or ad hoc conferences. This deployment covers:

Table 4.2 Overview of covered functionality

Prerequisites

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Before carrying out the configuration of Cisco VCS, Cisco MCU and Cisco TMS ensure that the

following prerequisites are met:

• At least one Cisco TelePresence VCS running X7.0 software.

• At least one MCU using 4.2 software for scheduled conferences.

• Additional MCU for ad hoc conferences.

• Cisco TMS running 13.1.2 software.

• Cisco VCS and TMS are installed and configured for base operation using the relevant

deployment guide (listed below).

• MCUs used start with the base settings.

• Cisco TMS has enough system licenses to add the relevant number of MCUs.

Step 1: Dial plan

The dial plan of a video deployment should be considered early on to ensure that a scalable easy-

to-use solution is deployed. This dial plan is a conceptual one that is not defined in any one place

but on a variety of systems: therefore it is important to follow the same guidelines throughout a

deployment. Recommendations that fulfill these core requirements are provided; however some

deployments may have specific requirements that require a different implementation.

Each conference has a numeric identifier. When a conference is booked using Cisco TMS, Cisco

TMS uses a pre-configured number range to create the conference. This registers numeric

identifiers on the Cisco VCS, so that participants can dial into the conference. For a scheduled

conference, Cisco TMS can configure the MCU to initiate calls to the participants (through the

Cisco VCS); this is most commonly done as a dial out call from the MCU to the endpoint.

All the conferences running on a specific MCU can be addressed using a number with a prefix

assigned from the address plan, for example: 81xxx, where 8 is the reserved prefix for data

centre resources and 1 is the prefix for a specific MCU. The same conference can also be reached

using a Unified Resource Identifier (URI), for example, [email protected], both on SIP and

H.323 (interworked) signaling protocols. (It is also possible to register multiple MCUs using the

same prefix in order to provide load balancing for ad hoc MCU conferences.)

Using a prefix allows a simplified dial plan where users need only dial <prefix><conference

number>@domain, whether using SIP or H.323.

The table below shows an example of an address plan for conferencing services. The range

allocated to ad hoc and permanent conferences can be divided as required. In both cases, a

conference address can be used across multiple sessions; for example, 81555 can be used for a

specific team’s shared meetings. Pre-registering a conference allows persistent tailoring of

layouts/settings across sessions.

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Overview of an address plan using five digits

Prefix/suffix Range Purpose Dialing examples 8 – Central

resources

1 – Cisco MCU/MCU pool number

000 -

010

Auto attendant calls

H.323: 81001

SIP: [email protected] or [email protected]

Interworked from H.323 -> SIP: [email protected]

011 -

909

Ad

hoc/preconfigu

red

conferences

H.323: 81123

SIP: [email protected] or [email protected]

Interworked from H.323 -> SIP: [email protected] 910 -

999

Reserved for Multiway

Never dialed directly

8 – Central

resources 2 – Cisco

MCU number

100-999 Scheduled

conferenc

es

Only for dial-in (TMS will make the

MCU dial out by default):

H.323: 82812

SIP [email protected] or

[email protected]

Step 2: Configuring the Cisco VCS The Cisco VCS Control should be deployed according to the recommendations of the Cisco VCS

Base configuration or the Unified CM with VCS deployment guide (both found at

http://www.cisco.com/en/US/partner/products/ps11337/products_installation_and_configuration

_guides_list.html). Configuring the Cisco VCS ready for the MCU installation requires the

following steps:

1. Configuring the MCU SIP sub domain.

2. Creating an MCU SIP zone.

3. Configuring search rules.

4. Optional: Configuring Multiway.

Configuring the MCU SIP domain The MCU registers to the Cisco VCS using a sub-domain, e.g. mcu1.cisco.com. Therefore, the

Cisco VCS has to be configured with a SIP domain name that matches the MCU sub-domain;

otherwise the Cisco VCS rejects the SIP registration request from the MCU.

Configure a SIP domain:

1. Go to VCS configuration > Protocols > SIP > Domains.

2. Click New.

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3. Enter the domain name into the Name field:

Settings for SIP domain

VCS Setting Value Comment Name

MCU fully qualified domain name (FQDN)

Example: mcu1.cisco.com or mcu1.cisco.net

4. Click Create domain.

Creating the MCU SIP zone To provide the same call behavior for SIP as for H.323, configure the Cisco VCS with a SIP

neighbor zone pointing to the MCU. (When using H.323, the MCU registers a service prefix; the

same does not exist for SIP.) Configure the neighbor zone with a pattern match equal to the

H.323 service prefix. To allow ad hoc calls to the MCU using a URI (for example, <conference

ID>@mcu1.cisco.com), configure the SIP zone with a suffix match with the pattern string

@mcu1.cisco.com.

This guide assumes that all video infrastructure devices that can be dialed use the 8 prefix

according to the address plan. The first MCU in a video network should then be assigned the

service prefix 1, thus giving the MCU prefix 81.

Create a SIP zone on the VCS as follows:

1. Go to VCS configuration > Zones.

2. Click New.

3. Configure the fields on the VCS as follows:

VCS Setting Value Comment Name

Zone name

Example: ToMCU1

Type Neighbor

Hop count 15

H.323 Mode Off

SIP Mode On

SIP Port 5061 If you don’t use encryption,

set this to 5060. SIP Transport TLS If you don’t use encryption,

set this to TCP. SIP TLS verify mode Configure the TLS

verification settings according to your security policy

Authentication policy Configure the

authentication settings

according to your

authentication policy

Refer to Authentication

Policy configuration

options in the VCS online

help for full details. Peer 1 address IP address or FQDN of

MCU

Example: mcu1.cisco.com

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Zone profile Infrastructure device

Configuring search rules

Search rules decide which calls will be routed to the MCU SIP zone.

Create a search rule on the VCS as follows:

1. Go to VCS configuration > Dial plan > Search rules.

2. Click New.

3. Configure the fields on the VCS as follows:

VCS Setting Value Comment

Rule name Descriptive name for the search rule

Example name: MCU1 zone – no domain

Description Description of the rule Example name: Search MCU1

zone for SIP conferences

Priority 50 The match priority must be the same as the local zone full URI

Source Any

Request must be authenticated Configure the authentication settings according to your authentication policy

Refer to Authentication Policy configuration options in the VCS online help for full details.

Mode Alias pattern match

Pattern type Regex

Pattern string <mcu service

prefix>(\d+)@.*

It is expected that Business

to Business calls will

require a full E.164 to dial,

e.g. [email protected]

Pattern behavior Replace

Replace string \1@<mcu-fqdn> Example: \[email protected] Note: Using the FQDN is critical

On successful match Stop

Target <Name of zone configured

above>

Example: mcu1

State Enabled

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5. Click Save.

This search rule will match SIP calls made using the full number with prefix and manipulate the

URI to what the MCU expects.

Example:

This matches the search rule for MCU1 which has prefix 80, but the MCU expects to receive a

call to conference [email protected]; therefore, the search rule makes this alteration before

passing the call to the MCU zone.

This rule allows the caller to dial the same number whether they use H.323 or SIP and also

allows for the automatic appending of the endpoint domain (which an endpoint will do if the user

does not specify a domain when they make a call).

Create another search rule on the VCS as follows:

1. Go to VCS configuration > Dial plan > Search rules.

2. Click New.

3. Configure the fields on the VCS as follows

VCS Setting Value Comment

Rule name Descriptive name for the

search rule

Example name: MCU1 zone –

SIP domain Description Description of the rule Example name: Search MCU1 zone for SIP conferences

Priority 50 The match priority must

be the same as the local

zone full URI Source Any

Request must be authenticated Configure the

authentication settings

according to your

authentication policy

Refer to Authentication

Policy configuration

options in the VCS online

help for full details. Mode Alias pattern match

Pattern type Suffix

Pattern string @<mcu-fqdn> Example: @mcu1.cisco.com

Pattern behavior Leave

On successful match Continue

Target zone Name of zone configured

above

Example: MCU1

State Enabled

4. Click Save.

This search rule matches SIP calls made using the domain of the MCU; this is the call string that

TMS will use for scheduled conferences, for example.

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Example:

This matches the search rule for MCU1 which has domain mcu1.cisco.com, but the MCU

expects to receive a URI in this format and so no alteration is made before the call is sent to the

MCU zone.