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200-001

Cisco Video Network Devices Exam

Study Guide

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.

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

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

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 Ciscosuch as switches, routers, gateways, session border controllers, and wireless access pointsare 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 organizations 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

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

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 realtime 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

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 servicesincluding virtualization, DHCP, DNS, various forms of security, and media resourcesthat 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 organizations 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, highdefinition 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 experienceend-toend. 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 MonitorAllows 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. MediatraceDiscovers 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 MetadataManages 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 organizations business objectives. The advantage of bringing collaboration technologies to the Cisco SBAvalidated 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 todays 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 organizations 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, todays 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 signagerepresenting 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 Managerthe Web-based central management application for all Cisco Digital Media System productsto 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 mediaincluding high-definition live broadcasts, on-demand video, Flash animations, text tickers, and other Web contenton 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 marketleading 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 Players 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 CaptureTransformShare 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 evergrowing 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 CaptureTransformShare 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 CaptureTransformShare Solution

By addressing these trends with CaptureTransformShare, 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 CaptureTransformShare

The transform piece in CaptureTransformShare 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 CaptureTransformShare 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 organizations 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 organizations 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 Ciscos 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 organizationsincluding banks, retailers, corporations, schools, hospitals, and public-sector groupsto 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 linesthe 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 formatsfrom 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 videoand a variety of other usage reportsduring 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 onpremises 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 Ciscos fully immersive three-screen solutions to the executives or team leaders office. Now, resolving problems, making faster decisions, or simply checking in face-to-faceis 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 moments 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 moments notice: Combining vivid 1080p30 video and superior audio, the Cisco TelePresence System 500 Series brings a lifelike, in-person experience to longdistance 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 32inch-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 industrys 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 subcluster 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)

52 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 laptops 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 installations 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 Video resolution is reduced. Possible Cause or Description 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. Action 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.

Multipoint calls are dropped.

Minimum is 16000; the Cisco recommended value is 32000. While in a multipoint meeting, 1. Check the Cisco Unified a CTS endpoint places a call CM configuration to on hold or an audio add-in determine if duplex joins the meeting. As a result, streaming mode is enabled. a call is dropped. 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 In some cases, the CTS 500 turn on automatically. 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. 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.

Document camera cannot be controlled by the CTS or the phone.

Incorrect configuration settings on the document camera or on Cisco Unified CM.

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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.

Cisco Unified IP Phone does not register but CTS registers fine.

Non-MIDlets Phone user interface (UI) does not show Cisco TelePresence-specific information, or does not refresh until a service button is pushed.

Contact TAC for assistance. Phone profile or Directory Log onto Cisco Unified CM Number (DN) not provisioned and make sure that the phone properly in Cisco Unified CM. profile and Directory Number (DN) are created and configured properly. Service or Authentication Log onto the Cisco Unified URL is not configured CM administration phone properly on the Cisco Unified device page and check the CM phone device page. following: 1. Service on the device. 2. Authentication and Idle URLVerify that this address points to the correct codec IP address, and follow the recommended format. 1. Check the cable connection from the primary unit to the Cisco

The phone does not display the Cisco TelePresence idle screen.

Phone may not be recognized:

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Cisco Unified CM Unified IP Phone. does not know about it. 2. Verify phone registration CTS is not registered by logging in to the Cisco because it is Unified CM administration unplugged. page. The phone did not a. Click on the IP address receive an IP address. and verify phone registration. There could be errors in b. Verify the phone in the the system. Cisco Unified CM Phone 3. Correct typos in the URL. Configuration window: Incorrect IP address. Typos in the external location URLs. 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.

Cisco Unified IP Phone does not register.

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. 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. 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 MIDlets fails to initialize properly, showing the following message: Configuration error.

CTS was moved to a different Cisco Unified CM and the registration is rejected.

The Cisco TelePresence unit does not register.

MIDlet cannot establish TCP connections with the CTS. Possible incorrect CTS IP address is assigned to the phone profile authentication server URL. The MIDlet does not find Authentication Server URL or it does not recognize its format. Non-English characters are used on phone device profile. 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. 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. 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. 1. Properly configure the Authentication Server URL for the phone. 2. Click Save. 3. Reset the phone.

Remove the CTL file from admin graphical user interface (GUI) in the Cisco Unified CM Administration interface.

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. Log onto Cisco Unified CM, make sure that the CTS profile and the directory number (DN) are created and configured

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.

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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. 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.

properly. Completely delete from Cisco Unified CM the CTS and the phone, including its associated DN, then add them back to Cisco Unified CM.

CTS un-registers from time to time.

The system experiences a SIP registration timeout. Intermittent network issues could cause packets to be dropped.

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

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). 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. Verify that the device or resource is configured properly in Cisco Unified CM.

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Time does not show correctly on the CTS or phone.

SNMP query does not work.

TelePresence Multipoint Switch (CTMS) 1080p conference. Network Time Protocol (NTP) 1. If NTP is not configured, is not configured properly or access Cisco Unified CM the codec does not sync up date/time group, configure with NTP. 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 is not configured 1. Make sure SNMP is properly in Cisco Unified CM configured properly in or the configuration is not Cisco Unified CM and that accepted by the CTS. 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 Action Description Cannot ping primary There is no DHCP Server 1. Connect your laptop to the codec. available on the network. camera port on the codec. Phone shows requesting The laptop should get a service message. The codec looks for DHCP by DHCP IP address Display shows the IP default during first-time assigned. address 192.168.100.2 bootup. If there is no DHCP 2. Use SSH to access the when codec boots up. available, the codec times out codec and restart calling or gets stuck in the boot up b. service by entering the

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sequence.

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. Unable to reach the secondary codec. Web UI shows a red X on the secondary codec.

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. The switch port is configured On the switch to which the for the same VLAN ID on codec is connected, configure both voice and access VLAN, a different VLAN ID for the which is not recommended. following: Voice Access VLAN If the primary codec cannot 1. Check the physical cable ping the secondary codec: connections. 2. On multi-screen systems, The secondary codec may reboot all three codecs and be down. verify that the correct IP There could be another address shows on all three hardware failure. main display screens upon reboot. If the primary codec can ping 3. If you are denied SSH the secondary codec, but the access to the codec primary codec still indicates through the camera port that the secondary codec is not using your administration reachable, there may be account, the problem may nonvolatile (NV) corruption or be NV corruption. Contact another software related issue. TAC for assistance. 4. If SSH access is granted, try a factory reset on the secondary codec. This may be hardware failure Contact TAC for assistance. or non-volatile (NV) corruption. 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

Upon CTS bootup, the system does not pass all six green checks, or gets stuck on one of them. CTS reboots by itself without user intervention.

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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 Possible memory leak or 1. If you can still access the through SSH or GUI gives system resource exhausted: system through SSH, error. Some process consumes enter the following CLI While the system responds majority of memory resources command to get a snapshot to regular ping requests, an or hog CPU resource, hence of the system state: internal server error system locks up. show tech runtime shown on the Web UI. 2. Reboot the codec if necessary to recover the system. The Web GUI denies Login issues: 1. Wait until boot or upgrade username/password login, but is complete. SSH access is accepted. The codec is in the process 2. Restart calling service using of booting up or upgrading. the admin CLI. Calling service is not See the Cisco TelePresence running. Administration There is password file Software Command corruption on the GUI References home page on access. 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 Possible duplicate IP address 1. Go to another computer dropping, users are kicked out on the network that kicks and do the following: of SSH or Web GUI session users out of codec periodically. connectivity. 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.

SNMP query does not work.

SNMP is not configured properly in Cisco Unified CM or the configuration is not accepted by the CTS.

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. 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. 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 Reboot the codec to clear up the error condition.

Cannot download logs from Web UI.

Not enough disk space.

Peripheral shows red X on Web GUI, sometimes triggering SNMP alarms. Status Details WindowMicrophone and Speaker status show false positives (green check-mark, when they should show red

Could be introduced by transient state changes on various CTS components or false alarms. 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 MCUs 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.

98 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. 2. 3. 4. 5. Go to System > Service Parameters. Select the relevant Unified CM Server. Select the Cisco CallManager (Active) as the service. Select Advanced to show advanced options. 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. 2. 3. 4. Go to Media Resources > Conference Bridge. Click Add New. Select Conference Bridge Type as Cisco TelePresence MCU. Enter relevant fields and click Save.

Step 4: On Unified CM, configuring a media resource group list 1. 2. 3. 4. 5. 6. 7. Go Media Resources > Media Resource Group. Click Add New. Choose a name and move the MCU(s) into the Selected Media Resources area. Click Save. Go to Media Resources > Media Resource Group List. Click Add New. Choose a name and move the created Media Resource Groups into the Selected Media Resource Groups area. 8. Click Save.

100 Step 5: On Unified CM, assigning a Media Resource Group List to a device 1. 2. 3. 4. Go to Device > Phone. Select a device. Choose the Media Resource Group List that you created earlier. 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 easyto-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, xxx@mcu1.cisco.com, 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 teams 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 8 Central 000 resources 010 1 Cisco MCU/MCU pool number 011 909 Purpose Auto attendant calls Dialing examples H.323: 81001 SIP: 001@mcu1.cisco.com or 81001@cisco.com Interworked from H.323 -> SIP: 81001@mcu1.cisco.com H.323: 81123 SIP: 123@mcu1.cisco.com or 81123@cisco.com Interworked from H.323 -> SIP: 123@mcu1.cisco.com Never dialed directly

Ad hoc/preconfigu red conferences Reserved for Multiway

910 999 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 812@mcu2.cisco.com or 82812@cisco.com 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. 2. 3. 4. Configuring the MCU SIP sub domain. Creating an MCU SIP zone. Configuring search rules. 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 Name 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 Name Type Hop count H.323 Mode SIP Mode SIP Port SIP Transport SIP TLS verify mode Authentication policy Peer 1 address Value Zone name Neighbor 15 Off On 5061 TLS Configure the TLS verification settings according to your security policy Configure the authentication settings according IP address to oryour FQDN of authentication policy MCU Comment Example: ToMCU1 Value MCU fully qualified domain name (FQDN) Comment Example: mcu1.cisco.com or mcu1.cisco.net

If you dont use encryption, set this If dont use encryption, to you 5060. set this to TCP. Refer to Authentication Policy configuration options in mcu1.cisco.com the VCS online Example: help for full details.

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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 Rule name Description Value Descriptive name for the search rule Description of the rule Comment Example name: MCU1 zone no domain Example name: Search MCU1 zone for SIP conferences The match priority must be the same as the local zone full URI Refer to Authentication Policy configuration options in the VCS online help for full details.

Priority

50

Any Source the Request must be authenticated Configure authentication settings according to your authentication policy Alias pattern match Mode Regex Pattern type <mcu service Pattern string prefix>(\d+)@.*

It is expected that Business to Business calls will require a full E.164 to dial, e.g. +1753810001@companyb.com

Pattern behavior Replace string

Replace \1@<mcu-fqdn>

Example: \1@mcu1.cisco.com Note: Using the FQDN is critical

On successful match Target

Stop <Name of zone configured Example: mcu1 above> Enabled

State

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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 1111@mcu1.cisco.com; 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 Rule name Description Priority Value Descriptive name for the search rule Description of the rule 50 Comment Example name: MCU1 zone SIP Example name: Search domainzone for SIP MCU1 conferences The match priority must be the same as the local zone full URI Refer to Authentication Policy configuration options in the VCS online help for full details. Example: @mcu1.cisco.com

Any Source Request must be authenticated Configure the authentication settings according to your Alias pattern match Mode authentication policy Suffix Pattern type @<mcu-fqdn> Pattern string Leave Pattern behavior Continue On successful match Name of zone configured Target zone above Enabled State 4. Click Save.

Example: MCU1

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.