Cisco Rich Media Conferencing

Revised: February 7, 2017

Conferencing is an essential component of any collaboration system, especially when serving remote users and/or a large user base. Cisco Rich Media Conferencing offers features such as instant, permanent, and scheduled audio and video conferencing, as well as content sharing.

Conference bridges provide the conferencing function. A conference bridge is a resource that joins multiple participants into a single call (audio or video). It can accept any number of connections for a given conference, up to the maximum capacity allowed for a single conference on that device. The output display for a given party shows all connected parties minus the viewer’s own input.

Cisco Rich Media Conferencing solutions utilize various infrastructures to provide audio and video conferencing capability and, in some cases, content sharing. The conferencing infrastructure can be Cisco Unified CM using software or DSP resources, Cisco TelePresence, or Cisco WebEx Collaboration Cloud, and this chapter covers the design details pertaining to each solution.

Cisco Rich Media Conferencing solutions are available as on-premises, cloud, or hybrid deployments. This allows an organization to integrate with the Collaboration solution in which they have already invested or, alternatively, to implement a service that is hosted "in the cloud." This is one of the more important distinctions between the various solutions, and it is the first decision point when determining which solution is the best fit for an organization.

Cisco WebEx Software as a Service (SaaS) offers a completely off-premises solution, while Cisco Collaboration Meeting Rooms (CMR) Hybrid is a hybrid solution with a mix of on-premises and off-premises equipment. Organizations that have deployed Cisco Collaboration System will benefit most from leveraging an on-premises solution. The later sections of this chapter provide more detailed deployment options for each conferencing solution.

Table 11-1 summarizes available solutions from an on-premises cloud perspective.

 

Table 11-1 On-Premises, Cloud, and Hybrid Capabilities of Cisco Collaborative Solutions

Solution
Audio
Video
Content Sharing
On-premises
Cloud
On-premises
Cloud
On-premises
Cloud

Cisco WebEx Meetings Server

Yes

No

Yes1

No

Yes

No

Cisco WebEx SaaS

No

Yes

No

Yes 1

No

Yes

Cisco CMR Premises

Yes

No

Yes

No

Yes

No

Cisco CMR Hybrid

Yes

Yes

Yes

Yes

Yes

Yes

Cisco WebEx Meeting Center Video Conferencing

No

Yes

No

Yes

No

Yes

1.Cisco WebEx webcam video only, and no support with standards-based video.

To provide a satisfactory end-user experience, careful planning and design should be done when deploying Cisco Conferencing solutions so that users are enabled with the conferencing functionality they require.

To aid in the design, this chapter starts with an introduction of the different types of conferences supported in the Cisco Conferencing solutions, followed by detailed discussions of the following main topics for each solution:

  • Architecture

This section introduces the main components of the Conferencing solution and describes its advantages as well as the different conferencing mechanisms available through the various components of a collaboration system. Supported deployment models, solutions, and recommendations are discussed here as well.

  • High availability

This section discusses best practices for designing a resilient Cisco Conferencing solution; it also contains guidance for redundancy and load balancing.

  • Capacity planning

This section provides best practices and design information related to capacity limits and scalability for the Cisco Conferencing solution.

  • Design considerations

This section discusses general recommendations and best practices for the Cisco Conferencing solution design.

This chapter contains discussions on the following Cisco Conferencing solutions:

  • Cisco WebEx Software as a Service, (SaaS)
  • Cisco WebEx Meetings Server – for private cloud
  • Cisco WebEx Meeting Center Video Conferencing
  • Cisco Collaboration Meeting Rooms (CMR) Premises
  • Cisco Collaboration Meeting Rooms (CMR) Hybrid

What’s New in This Chapter

This chapter has been updated with new support and updated designs for Cisco Collaboration System Release (CSR) 11. x. You should read this entire chapter before deploying Conferencing in your Cisco Collaboration System.

Table 11-2 lists the topics that are new in this chapter or that have changed significantly from previous releases of this document.

Table 11-2 New or Changed Information Since the Previous Release of This Document

New or Revised Topic
Described in:
Revision Date

Audio conferences

Cisco Unified CM Audio Conferencing

February 7, 2017

Licensing

Licensing

February 7, 2017

OnePlusN conferences

OnePlusN

February 7, 2017

Cisco Collaboration Meeting Rooms (CMR) Cloud has been renamed to Cisco WebEx Meeting Center Video Conferencing

All sections of this document

February 7, 2017

Multistream video

Multistream Video for Enhanced Layouts

June 14, 2106

Various updates for Cisco Collaboration Meeting Rooms (CMR)

Cisco Collaboration Meeting Rooms Premises

Cisco WebEx Meeting Center Video Conferencing

June 14, 2016

Licensing information

Licensing

January 19, 2016

Cisco Conference Now

Cisco Unified CM Audio Conferencing

June 15, 2015

Cisco Collaboration Meeting Rooms (CMR) Premises

Cisco Collaboration Meeting Rooms Premises

June 15, 2015

Cisco WebEx Software as a Service (SaaS)

Cisco WebEx Software as a Service

June 15, 2015

Cisco WebEx Meetings Server

Cisco WebEx Meetings Server

June 15, 2015

Cisco Collaboration Meeting Rooms (CMR) Hybrid

Cisco Collaboration Meeting Rooms Hybrid

June 15, 2015

Types of Conferences

The Cisco Rich Media Conferencing solution supports the following types of conferences:

  • Instant conference

An instant audio or video conference (also referred to as an ad hoc conference) is an impromptu conference. Instant conferences are not scheduled or arranged prior to the conference. For example, a point-to-point call escalated to a multipoint conference is considered to be an instant conference.

  • Permanent conference

Permanent conferences (also referred to as meet-me, static, or rendezvous conferences) are predefined addresses that allow conferencing without previous scheduling. The conference host shares the address with other users, who can call in to that address at any time.

Permanent conference resources are used on a first-come-first-served basis (non-assured). For a guaranteed conference resource (assured), scheduled conferences should be used.

  • Scheduled conference

A scheduled conference is started by its initiator through a scheduling management system called Cisco TelePresence Management Suite (TMS). Conferences are booked via Cisco TMS with a start and end time and optionally with a predefined set of participants.

Cisco Rich Media Conferencing consists of the conferencing solutions described below. The details pertaining to each solution are described in each individual section that follows.

This solution allows Unified CM to use its internal software component or external hardware digital signal processors (DSPs) as the resources to perform audio conferencing.

Cisco Collaboration Meeting Rooms (CMR) Premises is an on-premises conferencing solution. Each user has a personal room that can be used to conduct meetings. Users can manage items such as their conference name, layouts, and PIN from the Cisco TMS portal.

Cisco CMR Hybrid combines the on-premises video conference and the WebEx Meeting Center conference into a single meeting, which allows TelePresence and WebEx participants to join and share voice, video, and content. CMR Hybrid meetings can be either scheduled or non-scheduled.

Cisco WebEx Meeting Center Video Conferencing (formerly Cisco Collaboration Meeting Rooms (CMR) Cloud) is an alternate conferencing deployment model that does not require any on-premises conferencing resources or management infrastructure. It supports both scheduled and non-scheduled meetings as well as TelePresence, audio, and WebEx participants in a single call, all hosted in the cloud.

Where cloud-based web and audio conferencing is not suitable, it is possible to use the on-premises WebEx Meetings Server solution. This product offers a standalone audio, video, and collaboration web conferencing platform.

Cisco Unified CM Audio Conferencing

Cisco Unified CM supports audio conferences using any of the following methods:

Software Audio Conferencing

The software-based audio conference bridges are provided by the IP Voice Media Streaming Application on Unified CM. The application must be enabled on each individual node in a cluster. A software unicast conference bridge is a standard conference mixer that is capable of mixing G.711 audio streams and Cisco Wideband audio streams. Any combination of Wideband or G.711 a-law and mu-law streams may be connected to the same conference. The number of conferences that can be supported on a given configuration depends on the server where the conference bridge software is running and on what other functionality has been enabled for the application. However, 256 is the maximum number of audio streams for this type. With 256 streams, a software conference media resource can handle 256 users in a single conference, or the software conference media resource can handle up to 64 conferencing resources with four users per conference. If the Cisco IP Voice Media Streaming Application service runs on the same server as the Cisco CallManager Service, a software conference should not exceed the maximum limit of 48 participants.

The Cisco IP Voice Media Streaming Application is a resource that can also be used for several functions, and the design must consider all functions together (see Cisco IP Voice Media Streaming Application). Since the capabilities of the software audio conference bridge are limited, Cisco recommends using a software audio conference bridge only in centralized deployments or in deployments where the use of a G.711 codec is acceptable for instant and meet-me audio conferencing. It is also important to note that the use of a software audio conference bridge in Unified CM will result in a higher load on the system than otherwise would be present.

Hardware Audio Conferencing

Digital signal processors (DSPs) that are configured through Cisco IOS as conference resources load firmware into the DSPs that are specific to conferencing functionality only, and these DSPs cannot be used for any other media feature. Any Cisco PVDM hardware may be used simultaneously in a single chassis for voice termination but may not be used simultaneously for other media resource functionality. DSPs on PVDM hardware are configured individually as voice termination, conferencing, media termination, or transcoding, so that DSPs on a single PVDM may be used as different resource types. Allocate DSPs to voice termination first, then to other functionality as needed. The DSP resources for a conference are reserved during configuration, based on the profile attributes and irrespective of how many participants actually join.

Hardware audio conference bridges offer a wider range of capabilities and codec format support than the software conference bridges. Cisco recommends using hardware audio conference bridges where the enterprise requires a more versatile audio conference bridge and codec support for higher-complexity codecs such as G.729 to take advantage of bandwidth savings.

Built-in Bridge

Built-in bridge refer to the DSP resources that are hosted by one of the endpoints in the call. Certain Cisco IP Phones have an on-board DSP for the built-in bridge functionality. The IP phone built-in bridge is the only embedded audio resource in the Cisco Rich Media Conferencing architecture. The built-in bridge, however, has limited conference functionality and cannot be used to launch a full conference. The built-in bridge in the Cisco IP Phones allows a user to:

  • Join calls across different lines that the IP phone might have, and convert those calls into a conference hosted on the built-in bridge.
  • Barge into a call of a different endpoint that shares the line (if the call is not set to private), and convert the call into a conference hosted on the built-in bridge.
  • Start a silent recording or monitoring session from the endpoint that is engaged on a call, and fork the media generated and received by the phone invoking the feature.

The built-in bridge of the Cisco IP Phones can encode and decode G.711 and G.729 codec formats. However, once the codec for the call has been selected, the built-in bridge codec selection is locked and the phone will be unable to change the codec used. Therefore, the best practice is to carefully analyze the call flow in which the built-in bridge might be invoked to avoid call drops.

The built-in bridge can mix a maximum of two calls and can fork only one call (two streams).

Cisco Conference Now

Cisco Conference Now is a Unified CM native application that provides permanent conference capability similar to Meet-Me. This application is targeted for small business customers who require a basic audio conferencing solution. With Cisco Conference Now, user can simply call into a centralized number and enter the appropriate meeting ID and a host or attendee PIN when prompted by the voice-guided system to join the conference.
Figure 11-1 shows the Cisco Conference Now architecture along with the components involved. Conference Now allows both external and internal callers to join a conference by dialing the Conference Now IVR directory number, which is a centralized conference assistant number. An IVR device is used to guide and collect information from the caller to join the conference by playing announcements. The IVR is a media resource device that enables Unified CM to play prerecorded announcements (.wav files) to devices such as Cisco IP Phones and gateways.

Figure 11-1 Cisco Conference Now Architecture

 

The administrator can enable the Conference Now option for a user. If enabled, the user gets a meeting number and must configure a host PIN to start the meeting. Also, an optional attendee access code can be configured for attendees to join the meeting. Prior to the meeting, the conference host distributes the meeting number and the optional access code to all the participants. To start the meeting, the host dials into Conference Now and enters both the meeting number and the host PIN. To join the meeting, the attendees dial into Conference Now and enter the meeting number along with the optional attendee access code. If the attendee dials into the meeting before the host, the attendee will be placed on Music on Hold (MoH). Conference Now uses conference bridges configured in the media resource group (MRG) and media resource group list (MRGL) associated with the host's calling device to perform the conferencing function. Ensure that both the conference bridge and the IVR resources are available to Unified CM in order to use the Conference Now feature.

Using a conference bridge other than the software-based Cisco IP Voice Media Streaming Application (IPVMS) from Unified CM might not provide the conference party entry and exit tone. For the best user experience, we recommend using the software-based Cisco IPVMS conference bridge for Conference Now. For detail on conference party entry and exit tone support, refer to the latest version of the Feature Configuration Guide for Cisco Unified Communications Manager, available at

http://www.cisco.com/c/en/us/support/unified-communications/unified-communications-manager-callmanager/products-maintenance-guides-list.html

Consider the following points when implementing Cisco Conference Now:

  • The IVR supports Out of Band DTMF only. Use an MTP to convert any DTMF capability mismatch.
  • The IVR supports G.711 (a-law and mu-law), G.729, and Wide Band 256K. IPVMS supports G.711 and Wide Band 256K. For other codec support, use a transcoder.
  • Conference Now does not support any advanced functionality such as a roster list or muting and un-muting attendees.

Cisco Collaboration Meeting Rooms Premises

Cisco Collaboration Meeting Rooms (CMR) Premises utilizes the Cisco TelePresence infrastructure on-premises to provide business quality video and audio conferences as well as content sharing. Each user in the system can have a personal, always-on meeting room with an associated video address (DN and/or URI) for participants to dial-in and join the meeting. Users can personalize their own Cisco CMR experience with customizable features such as PINs, welcome screens, and layouts from their user portal.

The Cisco CMR Premises architecture enables rich conferencing collaboration capabilities for endpoints registered to Cisco Unified Communications Manager (Unified CM) as well as the ability to integrate business-to-business audio and video systems and legacy H.323 video systems interworked from Cisco Expressway to Unified CM. This architecture provides a rich feature set by relying on a variety of components in the conferencing solution. The following sections present an overview of those components and their roles in the CMR Premises solution.

Role of Cisco TelePresence Conductor

Cisco TelePresence Conductor manages the bridge resources for all conference types. TelePresence Conductor selects which bridge or bridge pools to use to host a specific conference, and it balances the conference load across the bridges in the defined pools. Unified CM is unaware of the individual bridges in the network and communicates only with the TelePresence Conductor.

Using TelePresence Conductor for conferences has several benefits, including:

  • Increased efficiency by allowing all conference types to use the same TelePresence Servers
  • Better user experience through advanced TelePresence Server features such as ActiveControl and dynamic optimization of resources
  • Simpler deployment options through provisioned CMRs
  • Centralized management of scheduled resources
  • Increased scalability of a conferencing solution
  • Optionally, centralized multiparty license management of TelePresence Server bridges (requires TelePresence Conductor XC 4.0 and TelePresence Server 4.2, or later releases)
  • Report generation on multiparty license usage and bridge resource utilization (requires Cisco TelePresence Conductor XC 4.1 or later releases)

Note Cisco TelePresence MCU does not support multiparty licensing. However, if TelePresence Conductor has multiparty licensing enabled, the TelePresence MCU can be added but it requires screen licenses to be installed and continues to operate as port-based licenses.


TelePresence Conductor optimizes TelePresence Server resources dynamically when the Optimize resources setting is enabled in the TelePresence Conductor conference template. This enforces maximum resource usage of a participant based on the maximum receive bandwidth advertised by the resources at conference join. This can reduce the amount of resources conference calls use and allows more concurrent connections to take place. For more information, see the TelePresence Server release notes available at

http://www.cisco.com/c/en/us/support/conferencing/telepresence-server/products-release-notes-list.html

Role of the TelePresence Server

Cisco TelePresence Server is a scalable videoconferencing bridge that offers flexible video, audio, and content-sharing capabilities for multiparty videoconferencing. It allows users to easily create, launch, and join meetings using standards-based video endpoints, mobile devices, Cisco WebEx clients, and third-party video endpoints. It works in conjunction with Cisco TelePresence Conductor to offer flexible, cost-efficient conferencing.

Its benefits include:

  • A consistent user experience across mobile, desktop, or room-based videoconferencing solutions
  • Multiple layouts, and views optimized for the capabilities of each device
  • Enhanced user experience with features including Cisco TelePresence ActivePresence screen layout, individual participant identifiers, and Cisco ClearPath
  • The ability to extend scale and reach to more participants to join meetings by extending meetings to WebEx users through CMR Hybrid

Cisco TelePresence Server is available as a virtualized application compatible with standard Cisco Unified Computing System (UCS) servers, or you can deploy it on dedicated hardware platforms. TelePresence Server operates in Remotely Managed mode only and requires TelePresence Conductor in the deployment.


Note TelePresence Server 4.3 and later releases support Remotely Managed mode only. Upgrading the software to release 4.3 automatically enables Remotely Managed mode.


Flexible licensing options are offered to enable you to deploy Cisco TelePresence Server capabilities in the way that best suits your needs. You can license Cisco TelePresence Server in conjunction with Cisco TelePresence Conductor on a per-user basis (multiparty licensing), or on a concurrent call basis (screen licenses) to enable the whole enterprise without restrictions.

Role of Cisco TelePresence Management Suite (TMS)

Cisco TMS provides scheduling, control, and management of Cisco TelePresence conferencing and media services infrastructure and endpoints. Cisco TMS integrates and searches directories and external information sources. It also coordinates with third-party calendars such as Microsoft Outlook so that users of those systems can book Cisco TelePresence and CMR Hybrid meetings. Cisco TMS also includes ready-to-run reports that provide enhanced TelePresence reporting and actionable insight for administrators.

The TelePresence Management Suite is a software application that runs virtualized on Cisco Unified Computing System (UCS) servers or on a customer-supplied server supporting medium to large networks. A maximum of 5,000 systems is supported; this includes endpoints, Cisco Unified CM, Multipoint Control Units (MCUs), and other infrastructure components.

The TelePresence Management Suite makes video services available to both administrators and conference organizers to schedule endpoints by using the following tools:

  • TelePresence Management Suite user interface — Provides complete control and advanced settings and is typically used by administrators.
  • TelePresence Management Suite Scheduler web application — Uses wizards to schedule conferences, add systems, and view availability. It is typically used by conference organizers.
  • Cisco TelePresence Management Suite Extension for Microsoft Exchange (TMSXE) — TMSXE is an extension of TMS that enables Cisco TelePresence scheduling through Microsoft Outlook. It does this by replicating Cisco TMS scheduled meetings to Microsoft Exchange room calendars. This extension enables conference organizers to set up conferences using their Microsoft Outlook client.
  • Cisco TelePresence Management Suite Extension for IBM Lotus Notes (TMSXN) — This extension enables conference organizers to set up conferences using their Lotus Notes client.
  • Cisco TelePresence Management Suite Extension Booking API (TMSBA) — This extension gives developers access to TMS booking functionality for custom integration with third-party calendaring applications. This enables conference organizers to set up conferences using their existing corporate calendaring interface.
  • Cisco TelePresence Management Suite Provisioning Extension (TMSPE) — TMSPE is an extension for TMS that enables rapid provisioning of TelePresence users, endpoints, and Collaboration Meeting Rooms (CMRs) for large-scale deployments. In conjunction with Cisco TelePresence Conductor, Cisco TMSPE provides an interface for the administrator to import and create users' personal CMRs as well as to provision multiparty licenses for users. It also hosts the TelePresence User Portal that allows users to customize their personal CMRs.

TMSXE, TMSXN, and TMSBA are optional plug-ins installed on the calendaring server to achieve calendar integration. Client machines do not have to be modified.

Conference Bridges for Non-Scheduled Conferences

For permanent and instant conferences, administrators group TelePresence Servers into pools in TelePresence Conductor, and TelePresence Conductor applies service preferences to prioritize the use of the pools for specific conference calls. These bridges are referred to as General TelePresence Server in the figures.

Conference Bridges for Scheduled Conferences

Scheduled conferences, including CMR Hybrid meetings for participation by Cisco WebEx users, utilize Cisco TelePresence Server bridges that are connected through TelePresence Conductor to host conferences. Figure 11-2 shows a dedicated pool of TelePresence Servers used for scheduled and Cisco WebEx conferences, where scheduling bridges are connected via SIP to TelePresence Conductor and Unified CM engages TelePresence Conductor via SIP to establish the video conferences.

Figure 11-2 Scheduling with TelePresence Conductor and Dedicated Resources

 

Figure 11-2 illustrates a design where TelePresence Conductor manages separate TelePresence Server pools; one pool is dedicated to scheduled conferences (Scheduled TelePresence Server in the illustration) while the other pool is dedicated to non-scheduled conferences (General TelePresence Server in the illustration). In this design, the resource availability for scheduled conferences might be guaranteed, but the conference resources from both pools might not be used very efficiently because scheduled conference resources might be idle a majority of the time while the non-scheduled conference resources are heavily used.

Figure 11-3 shows multiple shared TelePresence Servers used for both scheduled and non-scheduled conferences as well as Cisco WebEx conferences, and it is the current recommended design. In this design, the TelePresence Servers (General TelePresence Server in the illustration) are grouped together as a pool and used on a first-come-first-served basis by the TelePresence Conductor for all conference types.

Figure 11-3 Scheduling with TelePresence Conductor and Shared Resources

 

As shown in Figure 11-2 and Figure 11-3, conferencing resources can be dedicated for scheduled conferencing or they can be shared resources for both scheduled and non-scheduled conferences when deployed with TelePresence Conductor. There are some advantages and disadvantages of using either method (see Table 11-3 ):

  • Dedicated TelePresence Servers — Deploy one or more TelePresence Servers that are dedicated just for scheduled conferences, with each TelePresence Server in a separate bridge pool and service preference of its own. Optionally, a second bridge and pool combination can be used as a backup.
  • Shared TelePresence Servers — Allow TelePresence Servers to be used for non-scheduled as well as scheduled conferences. In this case, resource availability for scheduled conferences cannot be guaranteed because the necessary resources might already be in use by non-scheduled conferences. All TelePresence Servers can be configured into a single bridge pool if they are of identical size.

 

Table 11-3 Dedicated versus Shared TelePresence Server Resources

Type of Resource
Advantages
Disadvantages

Dedicated

By dedicating conferencing resources to scheduled conferences, resource availability is guaranteed, provided that enough resources are provisioned.

Takes up a TelePresence Server exclusively for scheduling. Non-scheduled conferences would need to use a different TelePresence Server.

Inefficient use of conferencing resources when users use more non-scheduled conferencing over a given period. More resources are necessary to handle the fluctuation in usage patterns.

No resource optimization takes place, and therefore non-scheduled conferences cannot use the available resources.

Cascaded conferencing cannot occur.

Shared

More efficient utilization of resources. When users use more or fewer scheduled resources, there is no impact on the number of resources left idle, as can happen with dedicated resources that are not used.

Cascaded conferencing is available (if enabled).

Targeted management of TelePresence Server resources is possible.

Non-scheduled conferences are able to reuse the resources freed up by resource optimization of scheduled conferences.

Over time, monitoring of resource usage patterns can identify the most appropriate pool configuration.

Conference resources are not guaranteed to be available for scheduled conferences because non-scheduled conferences could use all of the resources. To reduce this rick, adjust the value of the Capacity Adjustment parameter in Cisco TMS.

A number of different designs can be implemented, depending on the conferencing requirements of the desired solution. Some of those requirements include:

  • Scheduled versus non-scheduled conferencing
  • Resource availability – dedicated versus shared

For video conferencing, Cisco recommends using TelePresence Conductor for centralized management of the video conferencing solution with TelePresence Server as the conference bridge of choice for both audio-only and video conferencing, and any combination thereof achieved through screen licenses.

For more information about screen licenses per call type, including audio-only licensing, refer to the Cisco TelePresence Server Data Sheet available at

http://www.cisco.com/c/en/us/products/conferencing/telepresence-server/datasheet-listing.html

Scheduled Versus Non-Scheduled Conferencing

Scheduled conferencing utilizes a scheduling management system such as Cisco TelePresence Management Suite (TMS). Conferences are booked via Cisco TMS with a start and end time, and optionally with a predefined set of participants.

In this design, the solution supports scheduling on TMS with conferencing bridges engaged by the TelePresence Conductor. Options are available for scheduling with Cisco TMS, such as using Microsoft Outlook, through TMSXE, Smart Scheduler, and so forth. Scheduled conferences, including CMR Hybrid meetings for participation by Cisco WebEx users, utilize Cisco TelePresence Server bridges that are connected through TelePresence Conductor to host conferences. Figure 11-4 shows two dedicated TelePresence Servers used for scheduled and Cisco WebEx conferences connected to and managed by TelePresence Conductor. A SIP trunk is configured between Unified CM and TelePresence Conductor, and between TelePresence Conductor and the TelePresence Servers. In these scheduled conferencing call flows, Unified CM communicates directly with TelePresence Conductor to request the conferencing resources.


Note Conference bridge selection based on IP Zone will not function because TMS recognizes only IP Zones configured for TelePresence Conductor and not for individual bridges.


Figure 11-4 TMS Scheduling with TelePresence Conductor

 

Cisco TMS supports a two-node TelePresence Conductor cluster for failover. The two Conductor nodes must be added to Cisco TMS, and Cisco TMS selects one of the nodes as the primary. Cisco TMS polls both cluster peers every 5 seconds, and this poll interval is not configurable. If the primary cluster node fails to respond to the poll, TMS attempts two more polls before failing over to the peer node. During the failover, users can continue to book meetings, and TMS can start meetings and perform conference control without any manual intervention. Cisco TMS handles the automatic switchover to the available peer. When the failed primary cluster node becomes available again, Cisco TMS switches back to that primary cluster node.

Licensing

Cisco Multiparty is a user-based licensing model that is centrally managed by TelePresence Conductor, and it comes with two options: Personal and Shared. Personal Multiparty Plus (PMP+) is for specific named hosts, while Shared Multiparty Plus (SMP+) is for conference room systems or for sharing between users. Each license entitles a user to host a conference with unlimited participants and up to 1080p video resolution. Table 11-4 summarizes the features included in the Personal and Shared Multiparty licenses.

 

Table 11-4 Cisco Personal and Shared Multiparty Plus License Features

Feature
Personal Multiparty Plus
Shared Multiparty Plus

Tied to a named host

Yes

No

Availability

Included in Cisco UWL Meeting

A la carte or discounted with room system

Minimum order

25

1

Maximum conference size

Unrestricted, within the limit of available hardware capacity

Maximum resolution

1080p60 (full HD) for video and 1080p30 for content on single-screen or multi-screen endpoints

Rich media sessions for Business-to-business (B2B) or business-to-customer (B2C)

Included

Included

Cisco TelePresence Conductor, TMS, TMSXE, and Skype for Business/Lync Interoperability Licenses

Included

New customers buy with Starter Pack

Support for instant, personal CMR, and scheduled conferences

Yes

Yes

For more information on Multiparty licenses, refer to Cisco Multiparty Licensing At-a-Glance, available at

http://www.cisco.com/c/dam/en/us/solutions/collateral/collaboration/pervasive-conferencing/at-a-glance-c45-729835.pdf

Deployment Considerations

The physical location of a TelePresence Server is important to consider because media traffic flows between it and each participant in the conference. To provide the best experience for participants, centralize the location of the TelePresence Servers in each region where they will be deployed.

TelePresence Server always runs in remotely managed mode and requires TelePresence Conductor to operate. Figure 11-5 illustrates the deployment of TelePresence Server in remotely managed mode. TelePresence Conductor manages the TelePresence Servers via an XML-RPC API over HTTPs for all operations. TelePresence Conductor routes SIP signaling to the TelePresence Servers via its Back-to-Back User Agent (B2BUA).

TelePresence Server uses a secure connection for signaling and, optionally, secure RTP for media transmission. To use secure RTP between TelePresence Conductor and the TelePresence Server, the encryption feature key is required.

Figure 11-5 TelePresence Server in Remotely Managed Mode

 

Interworking H.323 Endpoints into a SIP Environment

Many designs require the ability to incorporate H.323 endpoints into the architecture and to ensure interworking with a SIP-based design. The Cisco Video Communication Server (VCS) can be used to interwork these H.323 endpoints into a SIP-based design to provide features such as H.323-to-SIP conversion for call control, H.239-to-BFCP conversion for content sharing, and H.235-to-SIP for SRTP conversion. Figure 11-6 illustrates a conferencing design with VCS interworking the H.323 endpoints with SIP to Unified CM to access the SIP-based conferencing solution.

Figure 11-6 Architecture Overview for H.323 Interworking

 

Design Considerations for Audio and Video Conferencing

The following sections provide general information and an explanation about the resources used in the conferencing architecture, as well as best practices for the conference types that use those resources.

Audio Conferencing

If you are deploying a large-scale video conferencing solution, Cisco recommends also deploying audio conferencing as part of that solution. The video conferencing solution requires TelePresence Conductor and TelePresence Server, which can also be used for audio conferencing. This reduces the complexity and total cost of ownership of the overall conferencing solution.


Note TelePresence Server 4.4 and later releases support join and leave audio notifications.


It is important to note that a video multipoint resource can be used for audio-only conferences, but audio-only conferencing resources cannot be used for the audio portion of video conferences.

Video Conferencing

When integrated with the Cisco CMR Premises architecture, video-capable endpoints provide the capability to conduct video conferences that function similar to audio conferences. Video conferences can be instant, permanent, or scheduled. This section discusses the following main topics:

Videoconferencing resources are hardware or software types, and currently the main difference between software and hardware video resources is capacity:

  • Software videoconferencing bridges

Software videoconferencing bridges process video and audio for the conference using just software.

  • Hardware videoconferencing bridges

Hardware videoconferencing bridges have hardware DSPs that are used for the video conferences, such as Cisco TelePresence MSE 8000 Series and Cisco TelePresence MCU 5300 Series.

Meeting Experience

The video portion of the conference can operate in one of the following meeting experience modes, depending on the conferencing device:

In addition, video conferencing can use any of the following methods to select the dominant speaker:

Full-Screen Voice Activation

Voice-activated conferences take in the audio and video streams of all the participants, decide which participant is the dominant speaker, and send only the dominant speaker's video stream back out to all other participants. The participants then see a full-screen image of the dominant speaker (and the current speaker sees the previous dominant speaker). The audio streams from the participants (four in the case of the Cisco TelePresence MCU and Cisco TelePresence Server) are mixed together, so everyone hears everyone else, but only the dominant speaker's video is displayed. This mode is optimal when one participant speaks most of the time, as with an instructor teaching or training a group. Speaker (segment) switching and room switching fall under this category.

Continuous Presence

Continuous-presence conferences display some or all of the participants together in a composite view. The view can display the participants in a variety of layouts. Each layout offers the ability to make one of the squares voice-activated, which is useful if there are more participants in the conference than there are squares to display them all in the composite view. For instance, if you are using a four-way view but there are five participants in the call, only four of them will be displayed at any given time. You can make one of the squares in this case voice-activated so that participants 5 and 6 will switch in and out of that square, depending on who is the dominant speaker. The participants displayed in the other three squares would be fixed, and all of the squares can be manipulated through the conference control web-based user interface, DTMF (in the case of the Cisco TelePresence MCU and Cisco TelePresence Server), and Far End Camera Control (FECC, in the case of the Cisco TelePresence Server).On the other hand, if you are using the "equal panels layout family," the layout would change to 3x3 when the sixth participant joins. The audio portion of the conference follows or tracks the dominant speaker. Continuous presence is more popular than voice switching, and it is optimal for conferences or discussions between speakers at various sites.

OnePlusN

OnePlusN conferences display the active speaker in a larger pane, with other participants in smaller panes. This layout grows or shrinks automatically as participants join or leave the conference. Based on the number of participants, the layout starts as "single" and then grows to onePlus5, onePlus7, onePlus9, and stays at onePlus12 for 12 or more participants. If the number of participants grows beyond 13, the participant video will get swapped in and out depending on who was the last active speaker.

Cisco ActivePresence

The Cisco ActivePresence capability of Cisco TelePresence Server enables the delivery of next-generation multipoint conferencing by offering a view of all attendees in a meeting while giving prominence to the active speaker. While the active speaker occupies most of the screen, an overlay of others in the call appears in the lower third of the screen. This maintains the immersive feel of the life-size main speakers while giving participants a more natural view of everyone else sitting around the virtual table.

Voice Activation Mode

Using this mode, the video conference bridge automatically selects the dominant speaker by determining which conference participant is speaking the loudest and the longest. To determine loudness, the MCU calculates the strength of the voice signal for each participant. As conditions change throughout the conversation, the MCU automatically selects a new dominant speaker and switches the video to display that participant. A hold timer prevents the video from switching too hastily. To become the dominant speaker, a participant has to speak for a specified number of seconds and be more dominant than all other participants.

Manual Selection of the Dominant Speaker

The dominant speaker might be selected through the MCU's web-based conference control user interface. A user with privileges to log onto the MCU's web page, highlights a participant and selects that person as the important or dominant speaker. This action disables voice activity detection, and the dominant speaker remains constant until the chairperson either selects a new dominant speaker or re-enables voice activation mode.

Automatic Participant List Cycling

With this method, the MCU is configured to cycle through the participant list automatically, one participant at a time. The MCU stays on each participant for a configured period of time and then switches to the next participant in the list. The conference controller (or chairperson) can turn this feature on and off (re-enable voice activation mode) from the web interface.

Multistream Video for Enhanced Layouts

CMR Premises supports multistream video technology, which allows improved conference layouts on multistream capable endpoints. When the multistream option is enabled on the endpoints, endpoints can send multiple copies of the same video stream, but at different resolutions, to the bridge; and the bridge forwards the requested video streams to the receiving endpoint, which can compose and display the video layout locally. For single-screen endpoints, the participants are displayed individually instead of picture-in-picture (PiP) across the top in content sharing mode (see Figure 11-7). Dual-screen endpoints can display the conference participants across both screens (see Figure 11-8). A single conference can have a mix of multistreamed and transcoded endpoints.

Figure 11-7 Multistream Experience with Single-Screen Endpoint

 

Figure 11-8 Multistream Experience with Dual-Screen Endpoint

 

The following guidelines apply to multistream video deployments:

  • Cisco TelePresence Server on a virtual machine, Cisco Multiparty Media 310/320, or Media 820 must run TelePresence software 4.2 or later release.
  • Multistream capable endpoints must run Collaboration Software 8.0 or later (Cisco TelePresence SX20, SX80, MX200 G2, MX300 G2, MX700, and MX800). By default, the multistream option is disabled and must be enabled on both the sending and receiving endpoints.
  • A mix of multistream and non-multistream endpoints can coexist in a single conference.
  • Multistream video is support with Cisco Unified CM 11.0(1a) SU1 and Cisco VCS X8.5.1 or later releases.
  • The iX protocol must be turned on end-to-end along the signal path, and it must be enabled in the call control system and TelePresence Conductor.
  • The SIP Max Incoming Message Size service parameter in Unified CM must be set to 15,000 bytes or higher.
  • Main video for the call should have a minimum bit rate of 500 kbps, otherwise multistream will be disabled.
  • Multistream is a SIP-only feature and is not supported with H.323 or interworked calls.

Instant Video Conferences

Instant video conferences can be accomplished using embedded video resources (MultiSite) or dedicated video resources. The method for initiating an instant conference varies according to the call control used to initiate it. Cisco Collaboration endpoints managed by Cisco Unified CM can initiate the conference through the use of Conf, Join, or cBarge keys or the Add function (for endpoints with the MultiSite functionality), endpoints managed by Cisco VCS can initiate the conference by making use of Multiway or MultiSite functionality.

This section discusses how instant conferences occur with embedded and dedicated resources.


Note For H.323 and SIP clients with built-in MCUs, Unified CM allows functionality of the endpoint’s built-in MCU only if the client is SIP.


MultiSite TM

Certain endpoints are capable of escalating point-to-point calls with two endpoints to conferences with three or more endpoints, without the need for an external dedicated device, and this is referred to as MultiSite TM. The conference created using MultiSite is considered instant because it usually happens without prior planning and scheduling (see Figure 11-9). An option key is required to unlock the MultiSite feature. MultiSite conferences use the embedded resources in the endpoint for the conference creation. Endpoints with MultiSite capability that have the key installed can invoke this conference type whether they are managed by Cisco Unified CM or Cisco VCS.

Figure 11-9 Point-to-Point Call Escalated to Instant Conference Using Embedded Resources (MultiSite)

 

Instant Conferences Using Dedicated Devices

While many video endpoints on the market today are incapable of hosting conferences themselves and require an additional device to handle mixing the multiple video and audio streams, key factors for selecting dedicated video resources over embedded resources are bandwidth usage centralization, scalability, and cost efficiency. These multipoint resources are shared by a number of endpoints and are capable of hosting many conferences at the same time. The method in which a dedicated resource is invoked depends on the endpoints and the call control device(s) involved. In the case of users utilizing devices that can and are registered natively to Cisco Unified CM, the users can initiate an instant video conference with dedicated resources by using the Conf, Join, or cBarge keys.

Figure 11-10 illustrates one example of an instant conference using an external resource.

Figure 11-10 Instant Conference with a Dedicated Conferencing Device

 

The Cisco TelePresence Server and MCU, used in conjunction with Cisco TelePresence Conductor, are the dedicated resources that enable instant calls for TelePresence endpoints controlled by Cisco Unified CM.

Permanent Video Conferences

Permanent conferences can be initiated in several ways depending on the call control used by the conference initiator: MCU-IVR dial-in, and preconfigured permanent alias video conference initiation. Figure 11-11 depicts an example of a permanent conference taking place.

The permanent alias for a video conference is always available after it has been preconfigured by the administrator and the user needs to dial the alias to join the conference.

Figure 11-11 Example of a Permanent Conference

 

 

The Cisco TelePresence multipoint devices may use different names for permanent conferences, referring to them as permanent conferences or static conferences.

IVR for Dial-in Conference

Dial-in conferences can optionally use an interactive voice response (IVR) system to prompt users to enter the conference ID and the password (if one is configured) of the conference they want to join. You can use either of the following types of IVRs with the Cisco MCUs:

  • The IVR built into the MCU
  • Cisco Unified IP IVR

The built-in IVR of the MCU has the following characteristics:

  • Can prompt to create a conference or join by conference ID
  • Can prompt for the password of the conference
  • Supports both in-band and out-of-band (H.245 alphanumeric) DTMF
  • Cannot be customized to provide more flexible menus or functionality

The only items that the user can customize are the recorded audio file that is played to the user and the logo at the top of the screen.

If you want to have a single dial-in number and then prompt the user for the conference ID, you can use Cisco Unified IP IVR in conjunction with the MCU. Cisco Unified IP IVR has the following characteristics:

  • Only applicable for Cisco Unified CM integrations
  • Can prompt for the conference ID and the password (among other things)
  • Supports only out-of-band DTMF

That is, the calling device must support an out-of-band DTMF method (such as H.245 alphanumeric) on H.323 devices. These out-of-band DTMF messages are then relayed by Unified CM to the Cisco IP IVR server. If the calling device supports only in-band DTMF tones, the Cisco IP IVR server will not recognize them and the calling device will be unable to enter the conference.

  • Can be highly customized to provide more flexible menus and other advanced functionality

Customizations can include such things as verifying the user's account against a back-end database before permitting that user to enter into the conference, or queuing the participants until the chairperson joins.


Note Because Cisco Unified IP IVR supports only out-of-band signaling, it will not work with endpoints that use in-band DTMF tones.


With Cisco Unified IP IVR, users dial a CTI route point that routes the call to the Cisco Unified IP IVR server instead of dialing a route pattern that routes directly to the MCU. After collecting the DTMF digits of the conference ID, the Cisco Unified IP IVR then transfers the call to the route pattern that routes the call to the MCU. This transfer operation requires that the calling device supports having its media channels closed and reopened to a new destination. For example, an H.323 video device that calls the Cisco Unified IP IVR will initially negotiate an audio channel to the Cisco Unified IP IVR server and then, after entering the appropriate DTMF digits, it will be transferred to the MCU, at which point Unified CM will invoke the Empty Capabilities Set (ECS) procedure to close the audio channel between the endpoint and the Cisco Unified IP IVR server and open new logical channels between the endpoint and the MCU. If the H.323 video endpoint does not support receiving an ECS from Unified CM, it will react by misbehaving or disconnecting the call.

Scheduled Video Conferences

When using TelePresence Conductor for scheduling, only SIP is supported. Because SIP is the protocol of choice in Unified CM, Cisco recommends registering any H.323 MCUs to a Cisco TelePresence Video Communication Server (VCS) as a gatekeeper, and configuring H.323-SIP interworking from the VCS to Unified CM to provide support for scheduled video conferences when only H.323 MCUs are available in the infrastructure.

Scheduled video conferences provide users with available conference resources at the conference start time. Scheduled conferences can be joined in a variety of ways, as Table 11-5 describes.

 

Table 11-5 Call Launch Options for Scheduled Video Conferences

Launch Method
Description

One Button to Push (OBTP)

Conference dial-in information is automatically displayed on endpoints that support OBTP. For systems that do not support OBTP, an email with conference information is sent to the conference owner to forward to the participants.

Automatic Connect

All endpoints are automatically connected at the specified date and time.

Manual Connect or Hosted

Conference cannot begin until a specific endpoint (usually the conference organizer's endpoint) connects. After this endpoint connects, the remaining endpoints are either automatically connected or allowed to dial in manually.

No Connect

For conferences managed by the Cisco TelePresence Management Suite (TMS), this option only reserves the endpoints and conference ports. The conference can be started by clicking Connect for the participants in TMS Conference Control Center.

Reservation

Reserves the endpoints but does not initiate any connections.

Scheduling attempts to ensure endpoint and port resource availability and provides convenient methods to connect to TelePresence conferences. Most organizations already use calendaring applications to schedule conferences. In this case, calendaring integration enables users to schedule conferences with their existing calendaring client. TelePresence deployments often include a large quantity of endpoints and different infrastructure components. Without a centralized management component, provisioning, monitoring, and resource allocation are difficult if not impossible. Management platforms greatly simplify these processes.

The Cisco TelePresence scheduling and management options you choose depend on the type of calendaring your organization uses, the type of TelePresence deployment selected or already implemented by your organization, and the requirements or preferences of your organization.

Scheduled meetings work by integrating TelePresence resources and endpoints with corporate calendaring applications (see Figure 11-12). Cisco TelePresence Management Suite (TMS) resides between endpoints and calendaring applications to locate the proper multipoint resource for each scheduled conference, and to provide resource reservation. Cisco recommends deploying scheduled conferencing with the TelePresence Management Suite and creating conferences by scheduling three or more endpoints.

Figure 11-12 Scheduled Conference Using Integration with a Calendaring Application

 

Security in Video Conferences

Unified CM supports secure conferencing with SIP MCU integration types. With secure conferencing, Unified CM uses HTTPS to communicate to the MCU for conference scheduling, it uses TLS for call signaling, and it uses SRTP for media payload encryption. However, the conference is secure only if all the participants' endpoints support video encryption.

Alternatively, Cisco VCS supports secure conferencing in environments with H.323 and SIP MCUs. Cisco VCS can also offer security interworking between H.235 and SRTP, thus making it better suited for deployments in which security is used with SIP and H.323 protocols.

For more information about secure conferencing, see the chapter on Cisco Collaboration Security.

Conferencing Resources

A conferencing resource is the entity that performs the media conferencing, multiplexing, or media switching functions for the conference. The actual entity that hosts a multipoint conference may reside within a video endpoint or be separate from the endpoint in a dedicated device whose resources are shared by many endpoints. In many customer environments both options can be deployed. It is also important to understand that the video conferencing functionality is achieved by media transcoding and media switching. This section provides overall guidance on the most appropriate uses of each of the following conferencing resources:

Dedicated Video Resources

Cisco has a wide range of dedicated devices for audio and video conferencing. The following devices are relevant to Cisco Video Collaboration endpoints:

The dedicated conferencing device mixes the audio and video streams from each video endpoint and transmits a single audio and video stream back to each endpoint. In the case of multi-screen endpoints, multiple audio and video streams may be sent and received by the conferencing device.

Using a dedicated device has the following benefits:

  • Greater scalability
  • Enhanced feature sets (auto attendants, scheduling, presenter modes, and so forth)
  • Higher quality user experience
  • Reduced cost compared to enabling embedded conferencing on large numbers of endpoints

Cisco TelePresence Server

The Cisco TelePresence Server is a transcoding multipoint device that is available as an appliance, a blade running in the Cisco MSE 8000 Series chassis, or a virtual machine running on Cisco supported virtualized platforms. The TelePresence Server can connect many video and audio devices using a variety of protocols, including SIP, H.323, and TIP; and it supports video resolution up to 1080p. Currently it is the only Cisco multipoint device that supports both TIP and non-TIP multi-screen systems. The TelePresence Server is capable of instant, permanent, and scheduled conferences. In certain cases when TelePresence Servers are managed by TelePresence Conductor, TelePresence Conductor optimizes conference resources dynamically by freeing up unused resources that were initially allocated to endpoints. Cisco recommends using Cisco TelePresence Servers for all video conferencing.

Depending on the model, Cisco TelePresence Servers can run as a cluster for redundancy or increased capacity. For more information about clustering with TelePresence Servers, refer to the Cisco TelePresence Server product documentation available at

http://www.cisco.com/c/en/us/support/conferencing/telepresence-server/products-maintenance-guides-list.html

Cisco TelePresence Multipoint Control Unit (MCU)

The Cisco TelePresence Multipoint Control Unit (MCU) portfolio offers flexibility for a variety of video deployments. The MCUs are designed for single-screen Cisco and non-Cisco video endpoints using standards-based H.323 and SIP call signaling. The user's experience can vary depending upon which of the many custom screen layouts is chosen for a particular conference. The MCU supports instant, permanent, and scheduled conferences. Depending on the model, Cisco MCUs can support resolutions from QCIF up to 1080p in 4:3 and 16:9 aspect ratios and can support the video modes listed in Table 11-6 .

A global configuration setting on the MCU enables one of these modes, which in turn affects the supported resolutions and capacity. Port count depends on which mode is enabled because HD, HD+, and Full HD settings require more hardware resources than the lower-resolution SD setting. Table 11-6 describes the differences between these options.

 

Table 11-6 MCU Port Mode Options

Mode
Description

Full HD

The number of 1080p30 or 720p60 (symmetric) video ports available. This allows the MCU to transmit and receive at these resolutions.

HD+

The number of 1080p30 or 720p60 (asymmetric) video ports available. This mode allows the MCU only to transmit at these resolutions. This mode is not available for Cisco TelePresence MCU 5300 Series.

HD

The number of 720p30 or w448p60 (symmetric) video ports available. This mode allows the MCU to transmit and receive at these resolutions.

SD

The number of standard definition (up to 448p) symmetric video ports available.

nHD

The number of w360p30 video ports available on Cisco TelePresence MCU MSE 8510 and Cisco TelePresence MCU 5300 Series only.

Cisco recommends using dedicated video resources for greater conference scalability and when a large number of endpoints will be deployed in the organization. Cisco TelePresence MCUs can be pooled together and managed by Cisco TelePresence Conductor for increased flexibility.

Feature Comparison

When deciding which multipoint device to deploy, it is important not only to understand their audio and video capabilities, but also to know which features are supported on each device. Table 11-7 summarizes feature support for the multipoint devices.

 

Table 11-7 Comparison of Features Across Multipoint Devices

Feature
TelePresence Server
MCU
Comments

SIP and H.323 support

Yes

Yes

TelePresence Multipoint Switch uses SIP for initial call setup, but then relies on TIP to negotiate media

TIP support

Yes

No

Allows multi-screen telepresence immersive endpoints to switch screens intelligently

CMR Hybrid

Yes

Yes

Integration with WebEx Cloud, allowing two-way video, two-way audio, and content sharing between video and WebEx participants

Individual transcoding

Yes

Yes

The ability to support full range of SD and HD resolutions

TelePresence Management Suite scheduling

Yes

Yes

Multipoint device can have conferences created and scheduled via TelePresence Management Suite

ActivePresence

Yes

No

While MCU does not support ActivePresence, it does have a similar layout (large window for active speaker and smaller PiPs for other endpoints)

Custom layouts

Yes

Yes

Ability for users to change the experience on their endpoint only during an active meeting

VIP or Important Mode

Yes

Yes

Ability to designate a particular endpoint as the VIP or Important person and have that person always shown regardless of active speaker

Director controls

No

No

Allows the mapping and locking of a specific endpoint source to a specific endpoint destination.

Clustering

Yes

Yes

Clustering on TelePresence Server or MCU requires TelePresence Server MSE 8710 or TelePresence MCU MSE 8510 blades on MSE 8000 Series chassis

Cascading

Yes

Yes

Ability to combine two separate conferences on two separate devices to increase overall scale

Lecture Mode

No

Yes

Identifies the "lecturer" based on active speaker. The MCU presents a different layout to the lecturer than the other participants see.

Content sharing

Yes

Yes

TelePresence supports TIP auto-collaborate channel, H.239, and SIP BFCP. MCU supports only H.239 and SIP BFCP.

Auto attendant

Yes

Yes

TelePresence Server requires this to be enabled on per-meeting basis via scheduling or management device

Resource Optimization

Yes

No

Ability for the conference bridge to return unused resources back to the pool (requires TelePresence Conductor)

Virtualization Support

Yes

No

TelePresence Server software can run on a virtualized platform, an appliance, or a blade inside a chassis.

Embedded Video Resources (MultiSiteTM)

Some endpoints can support embedded video conferencing (the simplest form of video conferencing) in which one video endpoint hosts two or more calls simultaneously, and this capability is called MultiSite TM. Additional endpoints connect to the host endpoint, which mixes together the video and audio streams from all the endpoints. Benefits of embedded conferencing include low initial cost and little or no configuration necessary by an administrator. However, some limitations include:

  • Limited scale. The host endpoint has to mix the audio and video from every other endpoint; therefore, the size of the conference is limited by that host endpoint's capacity.
  • Endpoints with this capability require more bandwidth than other endpoints.
  • Depending on the endpoint model, there might be limitations in video resolution, which can degrade the overall user experience when compared to calls hosted on a dedicated multipoint device.

For the reasons listed above, Cisco recommends using the MultiSite feature only when limited size instant conferences are needed in the organization. Careful analysis also should be done to weigh the cost benefits of the number of endpoints requiring the MultiSite key versus the benefits of a full MCU.

Table 11-8 indicates which devices are capable of MultiSite, and thus of hosting multipoint conferences.

 

Table 11-8 MultiSite Capable Devices and Codecs

Endpoint or Codec
MultiSite Maximum Resolution
Maximum Number of MultiSite Participants (includes host)

SX20 Codec

576p

4 video + 1 audio

SX80 Codec

720p

1080p

5 video + 1 audio

4 video + 1 audio

C40 Codec

576p

4 video + 1 audio

C60 Codec

720p

4 video + 1 audio

C90 Codec

1080p

4 video + 1 audio

MX200 G2 and MX300 G2

576p

4 video + 1 audio

MX700 and MX800

720p

1080p

5 video + 1 audio

4 video + 1 audio

EX90

720p

4 video + 1 audio

Cisco TelePresence Conductor

Cisco TelePresence Conductor works in conjunction with Cisco Unified CM or Cisco VCS to simplify conferencing and the management of multipoint devices. See Role of Cisco TelePresence Conductor, for more information about its role in the Collaboration Solution. TelePresence Conductor is able to manage multiple Cisco MCUs and TelePresence Servers for instant, permanent, and scheduled conferences.

TelePresence Conductor can group multipoint resources into pools, thus allowing an administrator to take an individual MCU out of service without impacting service availability. Additionally, unique conference templates can be tailored to meet each user's personal preferences for settings such as participant layouts and PINs. Figure 11-13 and Figure 11-14 illustrate the steps that take place when TelePresence Conductor is used as a conference resource for Unified CM.

Figure 11-13 Flow of an Instant Conference Call with Unified CM

 

Figure 11-14 Flow of an Permanent Conference Call with Unified CM

 

Once the steps in Figure 11-13 or Figure 11-14 are complete, the call is set up and media flows between the endpoint and the conference bridge.

In the case of MCU management, TelePresence Conductor can automatically cascade an active conference to a separate MCU to expand total capacity, and this is transparent to the users. This automatic cascade functionality is supported with either TelePresence Server or TelePresence MCU.

Because of its inherent high availability, Cisco TelePresence Conductor is well suited for organizations where video conferencing resiliency has a premium value and organizations with a large number of multipoint control units.

High Availability

Proper design of the CMR Premises infrastructure requires building a robust and redundant solution from the bottom up. By structuring the solution with redundancy (redundant media resources groups, redundant route groups, Cisco TelePresence Conductor, and redundant media resources), you can build a highly available, fault tolerant, and redundant solution. The following sections provide design guidance for high availability:

Media Resource Groups and Lists

When a user of a Cisco Collaboration endpoint that uses Cisco Unified CM as its call control activates the Conf, Join, or MeetMe softkey, Unified CM uses the Media Resource Manager to select conference bridges. Conference bridges or MCU resources are configured in the media resource groups (MRGs). The media resource group lists (MRGLs) specify a prioritized list of MRGs and can be associated with the endpoints. The Media Resource Manager uses MRGLs of the endpoints for selecting the conference bridge. How you group the resources is completely at your discretion, but Cisco recommends grouping the resources by using a logical modeling of the geographical placement whenever possible so that all endpoints at a given site use the conference bridges closest to them.

Cisco Unified CM has the Intelligent Bridge Selection feature, which provides a method for selecting conference resources based on the capabilities of the endpoints in the conference. If there are two or more video endpoints when the conference is initiated and a videoconferencing resource is available, Intelligent Bridge Selection chooses that resource for the conference. On the other hand, if no videoconferencing resource is available or if there are no video-capable endpoints in the conference, Intelligent Bridge Selection chooses an available audio resource for the conference. Intelligent Bridge Selection provides an added functionality to select secure conference bridges for secure conferences. However, secure conference bridge selection is dependent on device capabilities. Unified CM may decide to allocate secure conference bridges in lieu of video or audio conference bridges. Flexibility to change the behavior of the Intelligent Bridge Selection functionality is provided through service parameter configurations in Unified CM.

Intelligent Bridge Selection has the following advantages over other methods of conference bridge selection:

  • Conference bridge selection by conference type - either secure, video, or audio conferences
  • Simplified media resource configuration
  • Optimized use of MCU video ports that potentially would have been used for audio-only conferences with other methods of bridge selection

All the conference bridge resources and MCUs can be in one MRGL, and Intelligent Bridge Selection will then select the conference bridge based on the need to do just an audio conference or a video conference.

Unified CM also supports an alternate way of selecting conference bridges, which can be specified by service parameter configurations. In this mode, Unified CM applies the following criteria to select the conference bridge resource to use, in the order listed here:

1. The priority order in which the media resource groups (MRGs) are listed in the media resource group list (MRGL)

2. Within the selected MRG, the resource that has been used the least

If the MCU is placed at the top of the MRGL for the phone, the MCU will always be chosen even for audio-only conferences that do not involve any video-capable participants. In this scenario, the MCU resources might be wasted on audio-only conferences and not be available to satisfy the request for a video conference when it occurs. This mode, however, is not recommended because it removes the intelligence from Unified CM to select the right resource on every conference made and should be used only by administrators who are aware of the system-wide effects of this service parameter setting.

For further information about media resource design, see the chapter on Media Resources.


Note MeetMe conferences do not use the Intelligent Bridge Selection feature.


Route List and Route Groups

Route lists and route groups are common call routing mechanisms of reliability for calls that leave the Cisco Unified CM domain. For media resources integrated with Cisco Unified CM as a trunk, route lists and route groups should be used to achieve high availability if backup multipoint control units (MCUs) exist. Call admission control can be preserved by setting the locations of the media resources based on the trunk being used for the call.

To learn more about Route List and Route Group resiliency mechanisms, see the chapter on Dial Plan.

Redundancy with Cisco TelePresence Conductor

The Cisco TelePresence Conductor has the ability to be resilient in several ways:

  • Within the pool configuration
  • With the service preference
  • With clustering of Conductors
  • At the integration points with call control devices

The TelePresence Conductor has the ability to manage conferencing resources such as the TelePresence Server and TelePresence MCUs. Within its configuration it has the ability to group those resources into pools. These pools of conferencing resources contain similar types of devices: TelePresence MCUs must be in a pool with other TelePresence MCUs only, and TelePresence Servers must be in a pool with other TelePresence Servers. In addition, it is best practice to try to differentiate the MCUs by HD or SD to allow for more granularity of the video services. This granularity will allow the administrative staff to assign endpoints such as the Cisco Unified IP Phone 9971 to the SD resources while making the HD resources available for the high definition endpoints such as the Cisco TelePresence Series devices. In addition, all other multi-screen and immersive systems such as Cisco TelePresence MX and IX Series endpoints can be directed by the Conductor to use the TelePresence Server, which supports ActivePresence and TIP, to maintain the immersive experience for all participants.

The ability to group conferencing resources by pools enables the Conductor to add redundancy at the pool level. Pool-level redundancy is achieved by having more than one device of a given type in the pool. The Conductor will load-balance the conference placement among the bridges residing in the pool.

In addition, the Conductor has the ability to create an ordered list of pools, which is called a service preference. It is similar to a route list or media resource group list in Unified CM. At this level of configuration, the Conductor can use primary and secondary pools to create a redundant model of conferencing resources. For example, if the service preference has a list of MCU pools in this order: Pool 1 for the US and Pool 2 for EMEA (see Figure 11-15), then the Conductor will use all the resources in the US pool of devices first and, if needed, it can make a cascaded link between the MCU in Pool 1 and the MCU in Pool 2. The Conductor does this automatically as a part of its intelligent conferencing selection process.

Figure 11-15 Example of Service Preference Device Pools

 

The TelePresence Conductor can also be configured to use multiple Conductors for redundancy at the system level. This is achieved by clustering together multiple Conductors. A maximum of three TelePresence Conductors can be used in a clustered design. Cisco recommends at least two Conductors in all designs to ensure high availability of all the conferencing resources.

Clustering Conductors requires a low-latency connection with less than 30 ms round trip between the Conductor nodes. During the clustering process, the Conductors performs a database synchronization of all table entries, such as aliases, templates, service preferences, pools, and present state table of the conference resource ports. The initial clustering process has to have a primary Conductor that is used as the initial database for the cluster. Once the initial process is done, any Conductor in the cluster can update the database that is synchronized across all the Conductor nodes.

For more details on clustering, refer to the Cisco TelePresence Conductor deployment guides available at

http://www.cisco.com/en/US/products/ps11775/products_installation_and_configuration_guides_list.html

When integrating Unified CM with the Conductor cluster, redundancy is achieved by having multiple connections to Unified CM for the instant and permanent conference calls. When the Conductor is clustered in this design, it requires unique virtual IP addresses that are the termination points for the Unified CM conference bridge and SIP trunk. Because the virtual IP addresses are unique on each Conductor cluster node, this information cannot be replicated in the database synchronization process and needs to be configured by the administrator. Once this configuration is complete, Cisco recommends using different Conductor cluster nodes for the primary, secondary, or tertiary links to Unified CM.

For example, as illustrated in Figure 11-16, a Unified CM should be configured to use the virtual IP address of Conductor_1's instant conference configuration as the primary connection, and the secondary connection is set to the virtual IP address of the instant conference configuration on Conductor_2. For the permanent calls, the destination address of the primary SIP trunk should be Conductor_2's virtual IP address for the permanent conference configuration, and the secondary connection is set to the virtual IP address of the permanent conference configuration on Conductor_1.

Figure 11-16 Conferencing Redundancy with Unified CM Call Control

 


Note Additional redundancy can be achieved within Unified CM for instant and permanent conference calls. For more information, refer to the sections on Media Resource Groups and Lists, and Route List and Route Groups.


Capacity Planning

Capacity planning is critical for successful conferencing deployments. Given the many features and functions provided by the conferencing services as well as the many different types of media resources used as part of the architecture, it is important to size the conferencing infrastructure and its individual components to ensure they meet the capacity needs of a particular deployment.

This section provides information and best practices for sizing the media resources used in the conferencing architecture.

Sizing the Conferencing Resources

The purpose of sizing a conferencing deployment is to determine the number of required concurrent connections to the TelePresence Servers. Considerations include:

  • Geographical location — Each region served by Unified CM should have dedicated conferencing resources. For example, there could be one central location for the US where Unified CM, TelePresence Servers, and other servers are installed, and one central location for EMEA.
  • Preference for TelePresence Server platforms — Virtualized or non-virtualized
  • TelePresence Server platform capacities — For capacity details, refer to the latest version of the Cisco TelePresence Server release notes available at

http://www.cisco.com/c/en/us/support/conferencing/telepresence-server/products-release-notes-list.html

  • TelePresence Conductor platform capacities — For capacity details, refer to the latest version of the Cisco TelePresence Conductor release notes available at

http://www.cisco.com/c/en/us/support/conferencing/telepresence-conductor/products-release-notes-list.html

  • Type of conference — Audio and/or video; scheduled and/or non-scheduled
  • Conference video resolutions — Higher quality conferences use more resources
  • Large conference requirements — For example, all-hands meetings

Conference resources are generally dedicated to a region in order to keep as much of the conference media on the regional network; therefore, sizing can be considered on a region-by-region basis. To properly size the conferencing deployment, use the Cisco Collaboration Sizing Tool, available at

http://cucst.cloudapps.cisco.com/landing


Note Consult with your Cisco sales representative for assistance on sizing the conferencing resources for your particular environment.


Resource Allocation and Allocation Logic

A critical part of the conferencing design is to allocate the correct resources in the right place. However, to do this you first need to understand how the allocation logic works to allocate the right amount of resources in the corresponding location. Resource allocation also works differently in different call control applications.

The following information applies to conferences that are initiated or processed by endpoints controlled by Cisco Unified CM:

  • If an instant conference that does not use the MultiSite or built-in bridge functionality is initiated by an endpoint registered to Unified CM, the resource is allocated based on the media resource group and list assigned to the conference initiator. All other endpoints in the conference just join their streams to the resource selected by the conference initiator. (See Figure 11-17.)
  • If a conference is initiated by an endpoint registered to Unified CM, the resource is allocated based on the media resource group and list assigned to the conference initiator. All other endpoints in the conference just join their streams to the resource selected by the conference initiator. (See Figure 11-17.)
  • If a permanent video conference is created, the conference call (all participants' individual calls) will be processed based on the dial plan applicable to the call (permanent alias or number dialed) and off-loaded to the applicable trunk multipoint device.
  • If a scheduled conference is created, the conference call (all participants' individual calls) will be processed based on the dial plan applicable to the call and off-loaded to the applicable trunk multipoint device.
  • The resource allocation logic for scheduled conferencing lies in the scheduling platform resource selection algorithm.

In Figure 11-17, assume that the conference was initiated by endpoint A, which has its local resource as the first option in the media resource group list. Note how, for this particular conferencing example, the usage of the local resource is bandwidth intensive because there are more remote users than local participants in the conference, thus causing the remote users’ streams to traverse the WAN.

Figure 11-17 Conference Initiated by Endpoint A

 

For further information about the processing of media resources groups and lists, refer to the section on Media Resource Groups and Lists, and the chapter on Media Resources.

Scalability

Cisco recommends following the guidelines in this section to scale the design of your conferencing deployment.

Media Resource Groups and Lists

Make use of media resource groups and lists whenever possible to achieve the desired scalability for instant and MeetMe conferences in a Cisco Unified CM deployment. For example, see Figure 11-18, where two MCUs are used to double the instant video conferencing capacity of Device Pool 2 by adding the two containing media resource groups into the media resource group list used by the pool. For more information on media resource groups and lists, see the chapter on Media Resources.

Figure 11-18 Example of Media Resource Groups and Lists

 

Scaling Video Conferencing with TelePresence Conductor

Make use of Cisco TelePresence Conductor to scale video conferencing services. TelePresence Conductor offers orchestration for multiple resources so that they can be allocated as needed, and it can span conferences through more than one multipoint resource when its capacity is exceeded. Figure 11-19 depicts an example of how a single Conductor server can improve the scalability of a deployment for instant video conferencing. In this example the resource allocation starts when the user (1) initiates a conference on Cisco Unified CM. This in turn sends the request (2) to the Conductor (3). The Conductor determines which is the most appropriate resource to be utilized and creates the conference in the MCU (4). It then replies to Unified CM with the conference details (5), and Unified CM starts the conference signaling negotiations with the MCU (6). The media (RTP) flows point-to-point (directly) between the endpoint(s) and the MCU. TelePresence Conductor pools the resources, thus making all those collective resource available to Unified CM as if they were a single resource.

Figure 11-19 Example of Cisco TelePresence Conductor Allocating Conference Resources

 

For further information about Cisco TelePresence Conductor, see the section on Redundancy with Cisco TelePresence Conductor. The information and logic documented in that section also apply to increasing the scalability in a deployment.

Clustering and Cascading

Make use of clustering and/or cascading if your multipoint device supports it. (For the definition and functionality of cascading, see the section on Cascading.) Clustering multipoint blades in a Cisco TelePresence MSE 8000 series chassis has resource implications because it can triple or quadruple the amount of bandwidth that is required for that chassis. On the other hand, although cascading can be used to increase conference capacity while maintaining a distributed multipoint deployment, note that cascading can create an inconsistent meeting experience. Cascading is automated by Cisco TelePresence Conductor.

Table 11-9 summarizes information about which multipoint devices are capable clustering or cascading.

 

Table 11-9 Multipoint Device Support for Clustering and Cascading

Device
Clustering 2
Cascading

Cisco TelePresence Server

Yes

Yes, but SIP only

Cisco MCU

Yes3

Yes, but H.323 only

2.Requires Cisco TelePresence Servers running on physical hardware, MCU 5300 Series, or MCU MSE 8510 blades in an MSE 8000 Series chassis.

3. The MCU 5300 can be configured for clustering when configured in a stack.

Design Considerations

This section provides general guidance and recommendations for the conferencing design.

Cisco Rich Media Conferencing Deployment Models

This section provides general information about the various Cisco Rich Media Conferencing deployment models. It also examines where and when each deployment model is most effective and best used.

Multiple Sites with Centralized Resources

Centralized designs are recommended for voice and video deployments with few collaboration endpoints, and for larger deployments that extend over a limited geographical area.

For centralized deployments, Cisco recommends locating the multipoint device at a regional or headquarters campus site with the necessary WAN bandwidth available to each of the remote sites, as well as the necessary LAN bandwidth within the campus. In addition, the multipoint device should be centrally located, based on the geographical location of the endpoints, to prevent unnecessary latency caused by back-hauling calls to a site at the far edge of the network, although this might not be entirely possible due to the existing network layout.

Figure 11-20 depicts an example deployment of centralized resources with video endpoints located in Dallas, London, and San Jose. In this example the New York location was chosen as the geographically central location for the multipoint device, thus providing the least latency to the video users.

Figure 11-20 Multiple Sites with Centralized Resources

 

Multiple Sites with Distributed Resources

Cisco recommends a distributed configuration for large voice and video deployments spread across separate geographical regions. As the collaboration network grows, it is very advantageous to distribute multipoint devices to minimize latency and save bandwidth across expensive WAN links.

The Cisco TelePresence Server virtualized platforms and MCU appliances are suited for organizations moving from a centralized to a distributed deployment. As the collaboration network usage grows further, these deployments can evolve their standalone multipoint devices to chassis-based blades for increased scale and redundancy.

Figure 11-21 shows a distributed deployment with a multipoint device in New York providing multipoint services for North America, and a multipoint device in Paris providing multipoint services for Europe.

Figure 11-21 Multiple Sites with Distributed Resources

 

Cisco Unified CM Session Management Edition

For Cisco Unified CM Session Management Edition (SME) with leaf clusters that are geographically distributed in different regions, Cisco recommends using a dedicated TelePresence Conductor cluster per region with conference devices local to the leaf cluster, thus eliminating the use of WAN links for local multipoint calls. For Cisco Unified CM SME with leaf clusters within the same region, a single Conductor cluster may be shared, as illustrated in Figure 11-22.

Figure 11-22 Cisco Unified CM SME Deployment with All Leaf Clusters in a Single Region

 

Figure 11-22 shows an example of a Cisco Unified CM SME deployment where all the leaf clusters are in a single region. Within the region, the TelePresence Conductor cluster is trunked to the SME for permanent and scheduled conferences; this is to simplify the routing process and minimize the number of required SIP trunks. In this configuration, the leaf clusters direct the default route (SIP route pattern and numeric route pattern) to the SME, which sends the call to the Conductor. For instant conferences, each leaf cluster requires a connection to the Conductor because conferences are initiated from the leaf clusters.

Design Recommendations

This section provides general recommendations for conferencing deployments.

Latency

For an optimal and natural experience regardless of deployment model, multipoint devices should be located at sites with one-way network latency of less than 150 ms between the multipoint device and any endpoints, provisioned with adequate port capacity and provided with ample bandwidth for the number of provisioned conferencing ports. Bandwidth requirements vary depending on the required maximum call rate of endpoints and the number of endpoints connecting to the multipoint device. Provision based on the maximum bandwidth that a particular endpoint requires for the desired call rate and resolution. For details, refer to the specific endpoints' data sheets available on http://www.cisco.com.

Cascading

Cascading refers to the ability to bridge together conferences on two multipoint devices as one conference. This increases the maximum number of endpoints supported in a conference and can reduce bandwidth across links, depending on how the multipoint devices are deployed. Note that cascading conferences can create an inconsistent user experience, especially when features such as continuous presence are used by the endpoints. This is due to the remote conference appearing as just another endpoint on the local conference.

When making use of cascading for a large briefing conference type, it is a best practice to have all slave MCUs set to full-screen voice switched, and the master MCU set to the desired Continuous Presence layout. All the main speakers should be located in the master MCU so that they can be provided with the best experience.

MTP Used with a Conference Bridge

Media termination points (MTPs) are used in a conference call when one or more participant devices in the conference use RFC 2833 signaling format. When the conference feature is invoked, Unified CM allocates MTP resources for every conference participant device in the call that supports only RFC 2833. This is regardless of the DTMF capabilities of the conference bridge used.

Cisco WebEx Software as a Service

Cisco WebEx is a collaborative conferencing solution that does not require any hardware to be deployed on-site. All services (audio, video, and content sharing) are hosted in the Internet through the Cisco WebEx Collaboration Cloud. This is often referred to as software-as-a-service (SaaS). Meetings can be initiated and attended from anywhere, anytime, and do not require connectivity back into the enterprise. This section describes solution characteristics and provides design guidance for deploying WebEx SaaS.

Architecture

Cisco WebEx SaaS utilizes the Cisco WebEx Collaboration Cloud to deliver the conferencing solution to the customers. The Cisco WebEx Collaboration Cloud is a global network created with a carrier-class information switching architecture, and only Cisco Collaboration traffic flows over this network. Figure 11-23 shows the Cisco WebEx Collaboration Cloud architecture.

Figure 11-23 Cisco WebEx Collaboration Cloud Architecture

 

This network is purpose-built for real-time communications and has been specially formulated to minimize latency associated with TCP-layer flows. The network consists of application-specific multimedia switches at key peering points to handle rapid session traffic and to guarantee a high quality of service for WebEx meetings. These switches are housed in highly secure Cisco data centers interconnected via dedicated lines that circumvent the public internet. These data centers are located near the major internet access points to route meeting traffic around the globe securely and reliably. In addition to these large data centers housing major meeting nodes, Cisco deploys nodes around the world. The network is built on fully redundant clusters with Global Site Backup. These services and other facilities form part of the Cisco WebEx Collaboration Cloud Operational Support System.

Users can connect to a WebEx meeting using the meeting client running on the computer or mobile device. Once the connection is established, the WebEx Collaboration Cloud manages all synchronous real-time interactions that make up a WebEx meeting, as depicted in Figure 11-23. Users access WebEx applications via browsers through the WebEx Collaboration Cloud, which resides within the Web Zone. The Applications Program Interface (API) ties the WebEx applications to the switching platform in the Meeting Zone within the WebEx Collaboration Cloud core. Numerous clusters of interconnected and distributed collaboration switches, their associated databases, and the logical and physical network infrastructure make up the WebEx Collaboration Cloud core. Multi-layer security components and the WebEx Operational Support System encircle the network with an additional layer of protection.

The WebEx Collaboration Cloud delivers real-time traffic reliably using intelligent routing, Global Site Backup (GSB), and Global Server Load Balancing (GSLB). Based on the geographic location of WebEx meeting participants, the WebEx Collaboration Cloud determines the point of presence that offers the lowest latency and best performance. WebEx meeting hosts automatically get a backup site physically located in a geographically distant Cisco data center within the same region. In the unlikely event that the primary WebEx site becomes unavailable, GSB automatically switches all meeting activity to the backup site. GSLB is a load-balancing design that directs traffic to the least congested switch in the WebEx Collaboration Cloud in order to minimize the delays. Thus, if one meeting switch has congestion, traffic is directed to an alternate switch, resulting in faster screen updates and synchronization among participants, and a better meeting experience.

In the WebEx deployment model shown in Figure 11-24, all the content, voice, and video traffic from every client traverses the internet and is mixed and managed in the cloud at the WebEx data center. The WebEx data center is logically divided into the Meeting Zone and the Web Zone. The Web Zone is responsible for things that happen before and after a web meeting. It incorporates tasks such as scheduling, user management, billing, reporting, and streaming recordings. The Meeting Zone is responsible for switching the actual meeting once it is in progress between the endpoints.

Figure 11-24 WebEx Deployment

 

The Meeting Zone consists of two subsystems. Within the Meeting Zone there are collaboration bridges that switch meeting content. The multimedia platform is responsible for mixing all of the VoIP and video streams within a meeting. To join a WebEx session, an attendee first connects to the Web Zone. The Web Zone traffic flows only before or after the meeting, is relatively low bandwidth, and is mainly non-real time. The real-time meeting content share flows to and from the Meeting Zone and can be bandwidth intensive. Its real-time nature can place a heavy burden on enterprise access infrastructure. For further details regarding network traffic planning, see Capacity Planning.

Meeting Center uses the H.264 AVC/SVC codec to provide high-definition video for the conference. Higher network bandwidth is needed for those deployments. For further details regarding network traffic optimization for high-definition video, see Capacity Planning.

Each WebEx Meeting Center host has a Personal Room with a fixed customizable URL. The host can use his room to conduct meetings, and participants can enter the room using that fixed URL. Lobby management functions are available to the host, such as maintaining privacy by locking the room to prevent others from entering while the meeting is in progress.

Starting with Cisco WebEx Meeting Center version WBS30, Meeting Center can be integrated with Cisco Spark for post-meeting collaboration for any host who has both WebEx and Spark entitlement. After the meeting ends, the host is offered the option to create a Spark Room with all the invitees and attendees pre-populated. The host can then use this room to upload meeting minutes and perform further discussion and collaboration after the meeting. This capability applies to both scheduled and personal room meetings. For more information about Cisco Spark, refer to the Cisco Spark documentation at

http://support.ciscospark.com

For details on IM and Presence services delivered by WebEx Collaboration Cloud, see the chapter on Collaboration Instant Messaging and Presence.

Security

By default, all WebEx meeting data is encrypted using 128-bit SSL encryption between the client and Cisco's Collaboration Cloud. SSL accelerators within the cloud decrypt the content sharing information and send it to a WebEx conference bridge that processes the content and sends it back through an SSL accelerator, where it is re-encrypted and sent back to the attendees. All Web Zone and Meeting Zone traffic is encrypted using 128-bit SSL where SSL accelerators are used to off-load the SSL function from the Web and Meeting Zone servers.

After the meeting ends, no session data is retained in the WebEx cloud or an attendee's computer. Only two types of data are retained on a long-term basis: billing and reporting information and optionally network based recordings, both of which are accessible only to authorized enterprise users.

Some limited caching of meeting data is carried out within the Meeting Zone, and this is done to ensure that users with connectivity issues or who may be joining the meeting after the start time receive a current fully synchronized version of the meeting content.

Independent third parties are used to conduct external audits covering both commercial and governmental security requirements, to ensure the WebEx cloud maintains its adherence to documented security best practices. WebEx performs an annual SSAE 16 audit in accordance with standards established by the AICPA, conducted by Price Waterhouse Coopers. The controls audited against WebEx are based on ISO-27002 standards. This highly respected and recognized audit validates that WebEx services have been audited in-depth against control objectives and control activities (that often include controls over information technology and security related processes) with respect to handling and processing customer data.

For customers that require enhanced security, there is also an option to perform end-to-end 256 bit AES encryption for collaboration bridge and multimedia content so that traffic is never decrypted in the cloud. End-to-end encryption results in some lost features such as NBRs. For more information on enhanced WebEx security options, refer to the white paper Unleash the Power of Highly Secure, Real-Time Collaboration, available at

http://www.cisco.com/en/US/products/ps12584/prod_white_papers_list.html


Note End-to-end encryption options are available for Meeting Center and Support Center meetings without additional cost.


Starting with Cisco WebEx Meeting Center version WBS31R1, recordings are encrypted for all Cisco WebEx centers, and site administrators can enforce the use of passwords to access the network-based recordings. When the host schedules the meeting, he/she can require all attendees joining the meeting to sign in with Single Sign-On (SSO) authentication and can restrict the meeting to invited attendees only. With version WBS31R2, site administrators can enable an option to allow only authenticated attendees to enter a host's unlocked Personal Room, while the unauthenticated attendees must wait in the lobby until the host admits them to enter the room.

Scheduling

With respect to scheduling and initiating meetings, WebEx provides cloud-based web scheduling capability, but most organizations prefer to schedule from their corporate email system (Exchange, Lotus Notes, and so forth) or other enterprise applications. The WebEx Productivity Tools is a bundle of integrations with well known desktop tools incorporated into a single application. A WebEx administrator can control the specific integrations that are provided through the tool to their organization's user population. It can be downloaded and installed from the WebEx site, or it can be pushed out locally using standard desktop management tools. To learn more about WebEx Productivity Tools, refer to the information available at

https://www.webex.com/support/productivity-tools.html

User Profile

There are several options for creating WebEx user profiles for an organization in the cloud. Security considerations for the actual usernames and passwords, as well as for handling a large number of user accounts, should be considered. A WebEx administrator can create user profiles manually by bulk import of a CSV template or by a programmatic approach. A programmatic approach uses one or a combination of the WebEx APIs, URL, and XML, or a Federated SSO solution. The programmatic approach can be used by a customer portal, which is an application such as a CRM tool or a Learning Management System that integrates directly into WebEx. In addition, the user can sign up for an account from the company's WebEx site, and the user profile will be created after the request has been approved.

For integrating directly with an organization's LDAP directory, Federated SSO with Security Assertion Markup Language (SAML) is the preferred approach. For more information regarding Federated SSO, refer to the white papers and technical notes available at

https://developer.cisco.com/site/webex-developer/develop-test/sso/reference/

High Availability

The Cisco WebEx Collaboration Cloud has a very high level of redundancy built in and is managed by Cisco. It is designed for continuous service with a very robust cut-over to the redundant meeting nodes during outages. In addition to the primary WebEx site, every customer has a backup site physically located in a geographically distant WebEx data center within the same region. If a customer's primary site is unavailable, Global Site Backup (GSB) automatically moves all meeting activity to the backup site. Neither the hosts nor the participants notice that they are being redirected to the backup site. The GSB system facilitates continuous accessibility to WebEx meetings globally, and all attributes, address books, preferences, meeting schedules, and other real-time data are kept in sync between the primary and backup sites. Because of this synchronization, GSB provides redundancy and disaster recovery both before and after the meetings.

Cisco WebEx Cloud Connected Audio

Cisco WebEx Cloud Connected Audio (CCA) is an audio conferencing solution based on a hybrid deployment model that uses the on-premises IP telephony network to provide an integrated audio experience for an organization's WebEx meetings. WebEx CCA implements a SIP trunk connection from the organization's IP telephony network into the WebEx cloud infrastructure (see Figure 11-25). The audio conferencing traffic traverses through this SIP connection instead of the service provider PSTN connection and, thus, WebEx CCA provides significant savings on audio cost and maintains the same integrated and intuitive user experience as other WebEx audio options.

Figure 11-25 Cisco WebEx Cloud Connected Audio High-Level Design

 

As shown in Figure 11-25, a typical WebEx CCA high-level design consists of the on-premises IP telephony network and the WebEx cloud infrastructure that are connected via the dedicated IP Peering Connections provided by the customer. The on-premises IP telephony network consists of a Cisco Unified Communications Manager (Unified CM) cluster and Cisco Unified Border Element. Cisco Unified Border Elements are deployed in the WebEx cloud infrastructure and they mark the entry point for an organization's IP telephony network. The Cisco Unified Border Elements in the cloud and at the customer site communicate with each other via SIP. WebEx CCA requires the customer to have two IP Peering Connections that connect with different WebEx data centers residing in geographically separated locations for redundancy purpose. The redundant IP links are configured in active/standby mode. All conferencing audio traffic flows through the primary link and fails-over to the secondary link if the primary link goes down. WebEx CCA also requires the gateway routers to support Border Gateway Protocol (BGP) and Bidirectional Forwarding Detection (BFD) protocol. BGP and BFD offer a significant faster re-convergence time in the event of a network failure.


Note The WebEx data center equipment, audio bridge, and servers run over the shared infrastructure along with other customers in the WebEx CCA solution.


Cisco Unified CM has a SIP connection with the WebEx cloud through the Cisco Unified Border Element at the customer site to handle telephony signal. The conference dial-in number is owned by the customer and is terminated at the customer site. Call routing is handled at customer the site, call signaling and audio traffic is handled over the redundant IP peering connections, and call mixing is handled in the cloud. When users dial the conference number within the enterprise, Cisco Unified CM routes the call over the dedicated SIP trunk through the Cisco Unified Border Element to the WebEx cloud without traversing through the PSTN. When the conference users request callback, WebEx sends the call to the Cisco Unified Border Element at the customer site that routes it to the destination end-point. If the conference users reside outside of the enterprise network, calls are routed through the PSTN before terminating or after leaving the customer's IP telephony network. WebEx CCA supports only the G.711 audio codec, RFC 2833 DTMF, and SIP signaling.

WebEx CCA has the highly available and fully redundant architecture that is designed to ensure continuous service operation. Every major component has two instances in active and standby mode, backing up each other. There are two IP Peering Connections handled by two independent pairs of routers, two pairs of Cisco Unified Border Elements, and two audio conferencing bridges. If any of these components fails, its standby counterpart takes over. If the active peering link fails, the network will converge via the standby connection. All existing calls continue, but with a very brief interruption of the media flow. Cisco Unified Border Elements use the Out-of-Dialog OPTIONS ping mechanism to monitor the operational state of each other. Cisco Unified Border Elements at the customer site also monitor the Cisco Unified CM cluster using the Out-of-Dialog OPTIONS ping mechanism. Failure in responding to the ping results in removal of the unresponsive element from the dial-peer list of the sender, which commences routing all new calls via the standby instance. In case the active WebEx audio bridge fails, all calls associated with the bridge are terminated and the standby WebEx audio bridge is activated. WebEx will then prompt the users with a new number to connect to the newly activated bridge, which also re-dials all system-originated calls (callbacks) from before the failure.

Consider the following guidelines when deploying Cisco WebEx Cloud Connected Audio:

  • Cisco recommends using Cisco Unified CM 8.5 or later release with the WebEx CCA deployment.
  • Cisco recommends using a dedicated Cisco Unified Border Element for the WebEx CCA deployment to ensure a sound architecture and easy troubleshooting.
  • Cisco Unified Border Element can be deployed on either a Cisco Integrated Services Router (ISR) or an Aggregated Services Router (ASR), depending on the audio port capacity requirements.
  • Use an access control list (ACL) instead of packet inspection to restrict traffic in the firewall on the IP Peering link.
  • The system administrator must provide at least one toll and one toll-free number for guest dial-in.
  • If an audio codec other than G.711 is desired, use a transcoder to transcode the audio stream to G.711 before sending it to WebEx.
  • One Direct Inward Dialing (DID) Digital Number Identification Service (DNIS) must be passed to the WebEx cloud via the Cisco Unified Border Element for all conferencing numbers.

For more information on Cisco WebEx Cloud Connected Audio, refer to the documentation available at

http://www.cisco.com/go/cwcca

Capacity Planning

For a given customer, the actual number of concurrent meetings is essentially unlimited. Different WebEx conferencing types have different capacities with respect to number of attendees. For a detailed product comparison table, refer to the Cisco WebEx Web Conferencing Product Comparison, available at

http://www.cisco.com/en/US/prod/ps10352/product_comparison.html

Network Traffic Planning

With the increased traffic out to the internet, it is important to consider network traffic planning. When planning for network traffic, the way that users use WebEx will make quite a bit of difference in the amount of traffic generated by the meeting. For example, if attendees use native presentation sharing (where the document is loaded to the WebEx site prior to sharing), it generates far less data than if they share their desktops. For a large enterprise, this can be important to understand to ensure correct traffic engineering, especially at the choke points in the network, such as the Internet access points. A preliminary estimate should be made around the average number of meetings to be hosted during the busy hour, along with the average number of attendees. Then, depending on the type and characteristics of these meetings, some projections on bandwidth requirements can be made. For more information regarding network traffic planning, please see the Cisco WebEx Network Bandwidth white paper, available at

http://www.cisco.com/c/en/us/products/collateral/conferencing/webex-meeting-center/white_paper_c11-691351.html

Design Considerations

Observe the following design considerations when implementing a Cisco WebEx SaaS solution:

  • Collaborative meeting systems typically result in increased top-of-the-hour call processing loads. Cisco partners and employees have access to capacity planning tools with parameters specific to collaborative meetings to help calculate the capacity of the Cisco Unified Communications System for large configurations. Contact your Cisco partner or Cisco Systems Engineer (SE) for assistance with sizing of your system. For Cisco partners and employees, the Cisco Unified Communications Sizing Tool is available at http://cucst.cloudapps.cisco.com/landing.
  • All connections from WebEx clients are initiated out to the cloud. Typically, opening pinholes in network firewalls is not required as long as the firewalls allow intranet devices to initiate TCP connections to the Internet.
  • Provision sufficient bandwidth for conference video and data traffic. See Network Traffic Planning, for details.
  • Based upon business requirements, design decisions have to be made about the following:

User creation and authentication options (see User Profile, for details)

Meetings scheduling options (see Scheduling, for details)

  • Cisco WebEx SaaS uses the multi-layer security model, and security extends from the WebEx infrastructure to the organization and individual meeting layer. There are various security options available, and depending on the business requirements., an organization can implement different levels of security. For security options and considerations, refer to the white paper Unleash the Power of Highly Secure, Real-Time Collaboration, available at

http://www.cisco.com/en/US/products/ps12584/prod_white_papers_list.html

  • For more details on the various Cisco Collaboration client offerings and how they fit into Cisco conferencing solutions, see the chapter on Collaboration Endpoints.

Cisco WebEx Meetings Server

Cisco WebEx Meetings Server is a highly secure, fully virtualized, private cloud conferencing solution that combines audio, video, and web conferencing in a single solution. Cisco WebEx Meetings Server addresses the needs of today's companies by presenting a comprehensive conferencing solution with all the tools needed for increased employee productivity as well as support for more dynamic collaboration and flexible work styles. Existing customers can build on their investment in Cisco Unified Communications and extend their existing implementation of Cisco Unified Communications Manager to include conferencing using the SIP architecture. In addition, Cisco WebEx Meetings Server leverages many capabilities from Cisco Unified CM to perform its functions; for example:

  • Use the SIP trunk connection with Unified CM to conduct teleconferencing
  • Utilize Unified CM's SIP trunk secure connection support for secure conferencing
  • Integrate with legacy or third-party PBXs through Unified CM
  • Leverage Unified CM's dual stack (IPv4 and IPv6) capability to support IPv6

These capabilities are discussed in more detail in the following sections.

Architecture

Cisco WebEx Meetings Server is a fully virtualized, software-based solution that runs on Cisco Unified Computing System (UCS). It uses the virtual appliance technology for rapid deployment of services. Virtual appliance simplifies the task of managing the system. For example, using the hypervisor technology, system components can easily be moved around for maintenance, or system components can easily be rolled back to a working version if problem arises. The virtual appliance is distributed in the form of an industry standard format, Open Virtual Appliance (OVA). All the software components required to install WebEx Meetings Server are packaged inside the OVA. Traditionally, using an executable installer to install individual software components would take hours to deploy the software. However, using OVA can significantly reduce the amount of time required to deploy the software because all software components are pre-packaged inside the file. Thus, virtual appliance technology can help tremendously to reduce the deployment time for Cisco WebEx Meetings Server.

Figure 11-26 shows the high-level architecture for Cisco WebEx Meetings Server using the non-split horizon network topology. (For details on the non-split horizon network topologies, refer to the Cisco WebEx Meetings Server Planning Guide, available at http://www.cisco.com/en/US/products/ps12732/products_installation_and_configuration_guides_list.html.) Inside the virtual appliance, there could be one or more virtual machines (VMs) running. These are the administration, web, and media virtual machines. The administration and web virtual machines serve as the back-end processing for the administration and WebEx sites. These sites handle tasks that happen before and after the meeting, such as configuration, scheduling/joining meetings, and recording playback. The media virtual machine provides resource allocation, teleconference call control, and media processing (voice, video, and data) during the meeting. The number of virtual machines running inside the virtual appliance depends on the capacity desired and on whether high availability is needed. This provides various options for deployment size.

Figure 11-26 Cisco WebEx Meetings Server High-Level Architecture

 

Cisco WebEx Meetings Server offers the option of deploying the Internet Reverse Proxy (or edge servers) in the DMZ to facilitate external access. This option provides two advantages. First, all external participants can securely access the WebEx conferences from the internet without going through a VPN. Second, mobile users can join the meetings from a mobile device anywhere as long as there is internet connectivity. Note that the Internet Reverse Proxy is mandatory if mobile client access is enabled.

Internet Reverse Proxy is used to terminate all inbound traffic from the internet inside the DMZ. The content is then forwarded to the internal virtual machines through an encrypted Secure Socket Layer (SSL) or Transport Layer Security (TLS) tunnel. This encrypted tunnel is established by the internal virtual machines connecting outbound to the Internet Reverse Proxy. Therefore, there is no need to open TCP ports inbound from the DMZ to the internal network on the internal firewall. However, some outbound ports from the internal network need to be opened on the internal firewall to allow communication with the Internet Reverse Proxy in the DMZ.

All end-user sessions are 100% encrypted using industry standard Secure Socket Layer (SSL) and Transport Layer Security (TLS). All traffic between the virtual machines is sent over the secure channel. Federal Information Processing Standard (FIPS) encryption can also be turned on by a single policy setting, providing US Department of Defense (DoD) level security. Alternatively, the Internet Reverse Proxy can be deployed behind the internal firewall as shown in Figure 11-27.

Figure 11-27 Internet Reverse Proxy Behind the Internal Firewall

 

 

For security concerns, an organization would typically take several months to get approval in deploying a component inside the DMZ. Using this methodology, it could eliminate any DMZ components and bypass the approval process to get the WebEx Meetings Server deployment done quickly. All internet traffic (HTTP on port 80 and SSL on port 443) to the external firewall should be forwarded to the internal firewall. This will minimize the number of ports that need to be opened in the external and internal firewalls. However, placing the Internet Reverse Proxy inside the internal network implies that inbound internet traffic will terminate in the internal network. Although direct internet access to the internal network could be controlled by the firewalls, not all organizations allow terminating internet traffic directly on their internal network. Ensure that this deployment does not violate your organization's IT policy before choosing this option.

In a large enterprise deployment, an organization would require the Single Sign On (SSO) capability to allow end users to sign in using their corporate credentials. Cisco WebEx Meetings Server can connect to the corporate LDAP directory using the industry standard SAML 2.0 for SSO.


Note Cisco WebEx Meetings Server supports Meeting Center only.



Note Starting with Cisco WebEx Meetings Server 1.1, Cisco Jabber integrated with the Cisco Unified CM IM and Presence Service can be used to join or start meetings hosted on WebEx Meetings Server. For Cisco Jabber support details, refer to the Cisco WebEx Meetings Server System Requirements, available at http://www.cisco.com/en/US/products/ps12732/prod_installation_guides_list.html.


Cisco Unified CM Integration

Cisco WebEx Meetings Server support both Cisco Unified CM and Session Management Edition (SME). Cisco Unified CM is a central piece of the WebEx Meetings Server architecture that allows the following:

  • Attendees joining the teleconference by means of Cisco IP Phone or PSTN
  • Integration of legacy or third-party PBXs with Cisco WebEx Meetings Server

Cisco Unified CM integrates with WebEx Meetings Server by means of SIP trunks to provide inbound and callback call control. Customer can choose to turn on security and run Transport Layer Security (TLS) and Secured Real-time Transport Protocol (SRTP) over the SIP trunk connection. A SIP trunk is configured in Unified CM with a destination address of the Load Balancer in WebEx Meetings Server, and then a route pattern (match the call-in access number configured in WebEx Meetings Server) must be used to route calls via the SIP trunk. A second SIP trunk is configured in Unified CM with a destination address of the Application Server in WebEx Meetings Server, and then a SIP route pattern must be used to route calls via the SIP trunk. When an attendee dials the access number to join the meeting, the first SIP trunk is used to send the call. After the call is connected and the caller enters the meeting ID, the Load Balancer issues a SIP REFER to Unified CM to send the caller to the Application Server that hosts the meeting via the second SIP trunk.

The system administrator can configure a SIP trunk in WebEx Meetings Server that points to a Unified CM to perform callback. Attendees can provide a callback number and have the system out-dial the number to the attendees to join the bridge. In the case of attendees requesting callback, the WebEx Meetings Server sends the SIP request to Unified CM along with the callback number via the configured SIP trunk. It is imperative for Unified CM to be able to resolve all dial strings received from a callback request to join the meetings. Callbacks may also be disabled system-wide by means of site administration settings. Unified CM is in control of all toll restrictions to various countries or other numbers that most enterprises will block, because WebEx Meetings Server does not have any toll restriction blocking itself.

WebEx Meetings Server supports the bidirectional SIP OPTIONS ping mechanism. The ping response from the remote end indicates that the remote end is active and whether it is ready to accept calls. Based on the response, WebEx Meetings Server or Unified CM can determine whether to send calls on the current SIP trunk or look for an alternate SIP trunk (if configured) to send calls. Note that SIP OPTIONS ping is supported in Cisco Unified CM 8.5 and later releases. Due to this reason, Cisco recommends using a compatible Cisco Unified CM version that supports SIP OPTIONS ping for Cisco WebEx Meetings Server deployment. For the list of compatible Unified CM versions, refer to the compatibility matrix in the Cisco WebEx Meetings Server System Requirements, available at

http://www.cisco.com/en/US/products/ps12732/prod_installation_guides_list.html


Note Cisco WebEx Meetings Server supports SIP trunk connection with Cisco Unified CM only.


Legacy PBX Integration

Some organizations that have a legacy PBX and are not ready to fully migrate to a Cisco Unified Communications solution, might want to use Cisco WebEx Meetings Server with their system for conferencing. Cisco Unified CM can be used to bridge the legacy PBX and Cisco WebEx Meetings Server together. Cisco WebEx Meetings Server can see only Unified CM and does not even know the PBX is behind Unified CM. As long as Unified CM can interoperate with the organization's PBX, Cisco WebEx Meetings Server can integrate with the organization's PBX. This integration can provide several benefits:

  • Allow users in the legacy system to experience the new technology
  • Allow an organization to adopt the new technology gradually, at its own pace
  • Protect the customer's investment in existing technology while allowing them to migrate to Cisco technology gradually

For further details on PBX interoperability with Unified CM, refer to the documentation available at

http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805b561d.html

IPv6 Support

Cisco WebEx Meetings Server supports IPv4 only or dual stack (IPv4 and IPv6) addressing for telephony audio, while telephony signaling remains at IPv4. Audio streams can be IPv4, IPv6, or a mix of IPv4 and IPv6 in the same meeting. Cisco WebEx Meetings Server supports Alternate Network Address Types (ANAT) to enable both IPv4 and IPv6 media addressing in the Session Description Protocol (SDP) during the SIP Offer and Answer exchange on the SIP trunk with Unified CM to establish a media connection using the preferred addressing scheme.

Both IPv4 and IPv6 devices can be used for teleconferencing. With IPv6 devices, Cisco WebEx Meetings Server leverages Unified CM's capacity to translate the IPv6 signaling to IPv4 and transport it over a SIP trunk to the Cisco WebEx Meetings Server. With the telephony media addressing, Cisco WebEx Meetings Server can convert between IPv4 and IPv6. Therefore, Cisco WebEx Meetings Server can support IPv6 without any expensive MTP resources.

With ANAT, Cisco WebEx Meetings Server can support IPv6 telephony audio without the support of IPv6 telephony signaling. However, ANAT must be supported on both ends of the Unified CM SIP trunk. Be sure to enable ANAT on the Unified CM SIP trunk, otherwise there will be a failure to establish the call when attendees request callback or attempt to dial in.

If the WebEx Meetings Server has IPv6 enabled, ANAT headers will be included in the media offer. WebEx Meetings Server will always answer with ANAT headers if the media offer includes ANAT headers. The following paragraphs describe the media address version selection process between the IPv6-enabled WebEx Meetings Server and the dual-stack Unified CM using the ANAT header.

When WebEx Meetings Server sends a call to Unified CM, the SDP offer contains both IPv4 and IPv6 media addresses. If the called device is IPv6, Unified CM chooses IPv6 for the media connection and answers with the IPv6 media address in the SDP; if the called device is dual-stack, Unified CM uses the IP Addressing Mode Preference for Media parameter to determine the address version in the answer SDP. If the parameter is set to IPv6, then IPv6 will be used for the media connection.

When Unified CM sends a call to the WebEx Meetings Server through the SIP trunk, WebEx Meetings Server receives the SDP offer with an ANAT header. If the SDP offer contains both IPv6 and IPv4 media addresses, WebEx Meetings Server answers with the higher precedence address version specified in the ANAT header, which would be IPv6 in this case. If the SDP contains only an IPv6 address, WebEx Meeting Server answers with an IPv6 media address.

For information on deploying IPv6 in a Cisco Unified Communications system, refer to the latest version of Deploying IPv6 in Unified Communications Networks with Cisco Unified Communication Manager, available at

http://www.cisco.com/go/ucsrnd

High Availability

Cisco WebEx Meetings Server uses the N+1 redundancy scheme to ensure system availability in the event of component failures. High availability is achieved by adding a local, redundant system to the primary system within the same data center. At the system level, virtual machines and components inside run in active/active mode. If one component goes down, the system restarts the component. Status information is exchanged between system components. Using this status information, the system is able to distribute the requests evenly among the active components. Depending on the deployment size, the number of virtual machines in the backup or redundant system might or might not be the same as in the primary system.

In the high availability system, when the virtual machine hosting the meeting goes down, affected meeting clients will automatically reconnect to the available service within a short period of time. However, depending on the nature of the failure and which component has failure, not all clients and meetings would be affected. For descriptions of high availability system behavior after a component failure, refer to the latest version of the Cisco WebEx Meetings Server Administration Guide, available at

http://www.cisco.com/c/en/us/support/conferencing/webex-meetings-server/products-installation-guides-list.html

Virtual IP Address

Inside the high availability system, there is a second network interface in the active administration and Internet Reverse Proxy virtual machine that is configured with the virtual IP address. The administration and WebEx site URLs use this virtual IP address to access the administration and WebEx sites. In the event of failover, the virtual IP address is moved over to the new active virtual machine. Thus, it provides access redundancy to the administration and WebEx site.

Multiple Data Center Design

Cisco WebEx Meetings Server can be deployed in multiple data centers (up to maximum of 2) for geographic redundancy or disaster recovery. In this deployment, there are two WebEx Meetings Server systems with identical deployment size, one in each data center, that are joined together to form a single logical system running in active/active mode. The first system added to the multi-data center system is the primary, and the system that is added after that is the secondary. When the secondary system is added to the multi-data center system, all its global data are overwritten with the data from the primary system and only configuration parameters local to the data center are preserved. Refer to the Cisco WebEx Meetings Server Administration Guide for details on the types of data that will be overwritten and preserved. Within each data center, there are local Unified CM instances for handling teleconferencing. System status is exchanged, and information about users and meetings is synchronized across data center peers over an encrypted SSL link. Administrators use a single URL to manage the systems, and participants use a single URL or one set of dial-in numbers to join the meeting. When participant join a meeting via the client, the system automatically chooses the data center closest to the participant to host the meeting, and the meeting is cascaded across data centers.

In the event of failure, if one component goes down in the data center, the system restarts that component. If the whole data center goes down, the surviving data center takes over without any manual intervention, and the system still runs with full capacity. When this happens, affected meeting clients automatically reconnect to the service in the surviving data center within a short period of time. However, depending on the nature of the failure and state of the client, the recovery mechanism might be different and would follow the same behavior as the high availability system. For detail descriptions, refer to the latest version of the Cisco WebEx Meetings Server Administration Guide, available at

http://www.cisco.com/c/en/us/support/conferencing/webex-meetings-server/products-installation-guides-list.html

Consider the following information when using the multiple data center design:

  • Configure NTP in all data centers.
  • A multi-data center license is required for the WebEx Meetings Server system in each data center. Install the licenses onto the primary data center system before joining the data centers.
  • A deployment size of 50 users per system is not supported, but larger system sizes are supported.
  • Running a high availability system within the data center is not supported.
  • Deploy local Unified CM instances in each data center.
  • Joining the systems together will not increase the total system capacity.
  • Either both data centers or neither data center can have Internet Reverse Proxy deployed.

Capacity Planning

The capacity of WebEx Meetings Server depends on the platform of choice and the number of conferencing nodes running in the deployment. For capacity planning details, see the section on Collaborative Conferencing.

Storage Planning

If recording meetings is a requirement, sufficient disk space should be allocated on the Network Attached Storage (NAS) device to store the recordings. For disk space allocation detail, refer to the Meeting Recordings section in the Cisco WebEx Meetings Server Planning Guide, available at

http://www.cisco.com/en/US/products/ps12732/products_installation_and_configuration_guides_list.html

Network Traffic Planning

Network traffic planning for WebEx Meetings Server collaboration consists of the following elements:

  • Call control bandwidth

Call control bandwidth is extremely small but critical. Co-locating the WebEx Meetings Server with Unified CM helps protect against issues with call control. Remote locations need proper QoS provisioning to ensure reliable operation. Call control bandwidth is used for establishment of calls between WebEx Meetings Server and Unified CM, and the amount of bandwidth required for each call depends on how the attendees join the meeting. For an attendee dialing into the meeting, the call consumes approximately the same amount of bandwidth as making two SIP calls. For an attendee requesting callback, the call consumes approximately the same amount of bandwidth as making one SIP call. For details about call control bandwidth estimation for SIP calls and QoS provisioning, see the chapter on Network Infrastructure.

  • Real-Time Transport Protocol (RTP) traffic bandwidth

RTP traffic consists of voice and video traffic. Voice bandwidth calculations depend on the audio codec used by each device. (See the chapter on Network Infrastructure, for bandwidth consumption by codec type.) Video bandwidth can be calculated the same way as WebEx SaaS. (See Network Traffic Planning.)

  • Web collaboration bandwidth

Web collaboration bandwidth for WebEx Meetings Server can be estimated the same way as WebEx SasS. (See Network Traffic Planning.)

  • Multiple data center deployment

For proper operation and optimal user experience with this deployment, there are network requirements for maximum round-trip delay time (RTT) and minimum guaranteed bandwidth plus additional bandwidth for each cascaded meeting between data centers. For network requirement details, refer to the latest Cisco WebEx Meetings Server Planning Guide and System Requirements, available at

http://www.cisco.com/en/US/products/ps12732/products_installation_and_configuration_guides_list.html

Design Consideration

The following additional design considerations apply to WebEx Meetings Server deployments:

  • For scenarios where any WebEx Meetings Server components are separated by network firewalls, it is imperative to ensure the correct pinholes are opened for all required traffic.
  • Collaborative meeting systems typically result in increased top-of-the-hour call processing load. Capacity planning tools with specific parameters for WebEx Meetings Server are available to Cisco partners and employees to help calculate the capacity of the Cisco Unified Communications System for large configurations. Contact your Cisco partner or Cisco Systems Engineer (SE) for assistance with sizing of your system. For Cisco partners and employees, the Cisco Unified Communications Sizing Tool is available at http://cucst.cloudapps.cisco.com/landing.
  • Using Transport Layer Security (TLS) and Secured Real-time Transport Protocol (SRTP) have no effect to the WebEx Meetings Server capacity. However, using TLS and SRTP does have an impact on Cisco Unified CM capacity.
  • WebEx Meetings Server has no built-in line echo cancellation. Use an external device such as a Cisco Integrated Service Router (ISR) to provide echo cancellation functionality.
  • For more details on the various Cisco Collaboration client offerings and how they fit into Cisco conferencing solutions, see the chapter on Collaboration Endpoints.
  • Call admission control with WebEx Meetings Server is performed by Unified CM. With locations-based call admission control, Unified CM can control bandwidth to the WebEx Meetings Server system by placing the SIP trunk specific to WebEx Meetings Server in a location with a set amount of audio bandwidth allowed. Alternatively, Unified CM supports the use of Resource Reservation Protocol (RSVP), which can also provide call admission control. For further information regarding call admission control strategies, see the chapter on Bandwidth Management.
  • Cisco recommends marking both the audio streams and video streams from WebEx Meetings Server as AF41 (DSCP 0x22) to preserve lip-sync. These values are configurable in WebEx Meetings Server Administration.
  • Web conferencing traffic is encrypted in SSL and is always marked best-effort (DSCP 0x00).

Reference Document

For network requirements, network topology, deployment size options, and other deployment requirements and options for WebEx Meetings Server, refer to the Cisco WebEx Meetings Server Planning Guide, available at

http://www.cisco.com/en/US/products/ps12732/products_installation_and_configuration_guides_list.html

Cisco Collaboration Meeting Rooms Hybrid

Cisco Collaboration Meeting Rooms (CMR) Hybrid is a collaboration conferencing platform that combines the video experience of Cisco TelePresence Conferencing with the presentation experience of Cisco WebEx Meeting into a single meeting. Cisco WebEx and TelePresence are optimized to work with standards-based video endpoints and WebEx meeting clients. They help customers to extend the reach of the meetings and simplify the experience for all participants. Attendees on TelePresence endpoints and WebEx meeting clients can securely share two-way video, audio, and content among themselves. This platform brings together the user experiences from two conferencing systems and extends the collaboration to more users on more devices in more locations.

Cisco CMR Hybrid allows an organizer to schedule meetings using the familiar interface of Microsoft Outlook enabled by the WebEx Productivity Tools or with the Cisco TelePresence Management Suite (TMS). The host selects the participants, adds the preferred endpoints and the WebEx information, and sends the invitation to all attendees. Using the productivity tools, the attendees receive one meeting invitation with all the information about how to join through TelePresence or WebEx. The meetings can be launched using One Button To Push (OBTP) from the TelePresence endpoint, or Cisco TMS can automatically connect the endpoints with the meetings at the scheduled start time.

Architecture

As shown in Figure 11-28, the high-level architecture of Cisco CMR Hybrid consists of the enterprise collaboration network and the WebEx Cloud infrastructure that are connected through an IP connection. The enterprise collaboration network consists of Cisco Unified Communications Manager (Unified CM), Cisco Expressway-C and Expressway-E, TelePresence Bridge pools that are managed by TelePresence Conductor, and Cisco TelePresence Management Suite (TMS). Cisco Unified CM is the call processing platform that provides call routing and call control for the TelePresence endpoints within the enterprise. Cisco Expressway-C and Expressway-E route calls between the enterprise network and WebEx Cloud. Cisco Unified CM connects with Cisco Expressway-C and Cisco TelePresence Conductor over separate Best Effort Early Offer SIP trunks.

For details on integrating Cisco Unified CM with Cisco Expressway, refer to the latest version of the Cisco Expressway and CUCM via SIP Trunk Deployment Guide, available at

http://www.cisco.com/c/en/us/support/unified-communications/expressway-series/products-installation-and-configuration-guides-list.html


Note For existing Cisco VCS customers, deployment using Cisco VCS Control and Expressway in place of Cisco Expressway-C and Expressway-E is supported.



Note Deployment using a Best Effort Early Offer SIP trunk between Unified CM and the TelePresence Bridge without TelePresence Conductor is supported, but using TelePresence Conductor is recommended.


Cisco TelePresence Conductor selects a TelePresence Bridge from the pool to host the TelePresence conference. The TelePresence Bridge mixes the audio from the TelePresence endpoint participants and sends the mixed audio, the active speaker video, and the content sharing video to the WebEx Cloud using SIP. Similarly, the TelePresence Bridge receives the media (mixed audio, active speaker, and content sharing video) from the WebEx Cloud, cascades the audio into the TelePresence conference, and sends the content sharing video to the TelePresence endpoints. If the TelePresence Bridge detects that the active speaker is from the WebEx side, it switches the TelePresence endpoints to the active speaker video. If the active speaker is from the TelePresence side, the TelePresence Bridge sends the previous active speaker video to the TelePresence endpoint of the current active speaker.

Figure 11-28 Cisco CMR Hybrid Using WebEx Audio with SIP

 

In the DMZ, Cisco Expressway-E handles the traversal calls between the enterprise and WebEx Cloud, and it allows the signal and media to traverse through the internal and external firewalls. Cisco Expressway-E connects with the WebEx Cloud through the configured DNS Zone and routes calls to WebEx via DNS lookup. Cisco Expressway-E communicates with WebEx Cloud via an encrypted connection using TLS and secured RTP for SIP signal and media. Customers have an option to turn on encryption for the SIP signal and media traffic within the enterprise. TelePresence endpoints outside of the enterprise can register with Unified CM through Expressway-C and Expressway-E, and thus participants on these endpoints can join the CMR Hybrid meetings.

When the WebEx Cloud receives the traversal calls and media sent from the enterprise network, the WebEx audio bridge cascades the audio into the WebEx conference, and WebEx switches to the active speaker video and displays the content sharing on the WebEx meeting clients. Similarly, WebEx Cloud sends the conference mixed audio, the active speaker, and content sharing video from the WebEx side to the enterprise via Cisco Expressway-E and Expressway-C, which routes them to the TelePresence Bridge.

Cisco CMR Hybrid supports H.264 video for active speaker and content sharing. It utilizes Binary Floor Control Protocol (BFCP) for content sharing and G.711 codec for audio. While Cisco WebEx uses H.264 video and G.711 audio codec, TelePresence can still use other video formats or codecs that are supported by the endpoints. The TelePresence Bridge will handle the audio and video interoperability between the TelePresence endpoints and WebEx meeting clients. In addition, there is a flow control on the link between the TelePresence Bridge and WebEx Cloud that regulates the bandwidth available for handling the media. For media from WebEx, the TelePresence Bridge always allocates 4 Mbps to ensure that WebEx sends the best quality of video possible to the TelePresence Bridge. For media from the TelePresence Bridge, the WebEx meeting client has a video floor of 180p for active speaker video at the minimum bit rate of 1.2 Mbps. If the minimum bit rate cannot be maintained due to network conditions (severe packets loss, for example), the WebEx client will stop receiving the active speaker video but still receives content sharing as well as conference audio and sends its video to other participants. Starting with WBS 29.11, the WebEx client will periodically perform bandwidth retest and automatically reestablish active speaker video when network conditions stabilize. Depending on the capability of the device running the WebEx meeting client and on bandwidth available, the WebEx client supports active speaker video up to HD 720p at 30 frames per second (fps) and content video up to 1080p. During the meeting, WebEx allocates the bandwidth based upon the least capable device among all WebEx clients in the conference (excluding devices running below the video floor), with a maximum bandwidth of 4 Mbps. However, if the least capable device leaves the conference, the bandwidth will be re-allocated based on the next least capable device that runs the WebEx meeting client. The allocated bandwidth determines the resolution and frame rates used to display TelePresence video on WebEx clients. Depending on the TelePresence endpoints deployed, video resolution required, screen layout desired, and deployment options chosen, customers can deploy the TelePresence Bridge using the Cisco TelePresence Server (appliance or virtualized platforms) or Cisco TelePresence MCU, but the pool must consists of bridges of the same type only (either TelePresence Server or TelePresence MCU). For TelePresence Conductor deployment details, see to the section on Cisco Collaboration Meeting Rooms Premises.

WebEx and TelePresence participants can join the CMR Hybrid meeting from within the enterprise or anywhere from the internet. For WebEx participants, they join the meeting using the WebEx meeting clients with either PSTN or VoIP audio. For TelePresence participants, they join the meeting via the One Button To Push (OBTP) or Auto Connect feature with the supported endpoints or by calling directly into the TelePresence Bridge. Once the participants successfully join the meeting, they can see the live video of each other from the endpoints and meeting clients. For presentation sharing with a WebEx user, either the user can make himself the presenter or the host can assign the presenter privilege to the user before he can start sharing the presentation. There is the WebEx site configuration to control this behavior. For presentation sharing with a TelePresence user, the user can connect the video display cable to his computer or press a button on the endpoint to start sharing his presentation without involving the host.


Note Staring with Cisco TMS 14.6 and TMSPE 1.4, Cisco Collaboration Meeting Rooms Premises can be integrated with Cisco WebEx, allowing participants to join a meeting in the user's personal room from the WebEx meeting client.


Scheduling

Cisco TelePresence Management Suite (TMS) is the key component for scheduling Cisco CMR Hybrid meetings. It provides a control link to the Cisco WebEx meeting scheduler. This link enables Cisco TMS to create new meetings on Cisco WebEx calendar and to obtain Cisco WebEx meeting information that is distributed to meeting participants. The following options are available to schedule CMR Hybrid meetings:

  • WebEx Productivity Tools

WebEx Productivity Tools is a suite of tools that allows users to schedule WebEx sessions quickly and easily. Productivity Tools includes an Outlook plug-in that allows an organizer to schedule WebEx Meetings, TelePresence resources, and CMR Hybrid meetings. Cisco TelePresence Management Suite Extension for Microsoft Exchange (TMSXE) is required for the productivity tool to interface with Cisco TMS for booking the meetings. This option provides a seamless integration for users to schedule CMR Hybrid meetings and to send the invitations to all participants directly inside the email client with a single transaction.

  • Smart Scheduler

Smart Scheduler is a web-based tool that is hosted on Cisco TelePresence Management Suite Provisioning Extension (TMSPE), and it allow users to schedule CMR Hybrid meetings using a browser. This could provide an option for users who would like to schedule meetings on mobile devices.


Note As long as the Cisco TMSPE option key has been installed, there is no extra license required for using Smart Scheduler.


  • WebEx Scheduling Mailbox

In this option, the network administrator needs to create a special mailbox account in Microsoft Exchange Server. When an organizer schedules a CMR Hybrid meeting, he should include this special mailbox account in the invitees list. Cisco TMSXE monitors this account and requests Cisco TMS to book a CMR Hybrid meeting if it sees this account in the recipients list. This option provides a convenient way, but with limited control of settings, for users to schedule meetings using any email clients that are supported by Exchange, such as Outlook Web Access (OWA).

  • Cisco TMS Booking Interface

With this option, the meeting organizer has to log in to the Cisco TMS portal and schedule the CMR Hybrid meetings from the Booking interface. This interface provides users with control of advanced settings for the meetings, and typically IT or help desk personnel uses this option to schedule meetings.

For Cisco TMS configuration details with these options, refer to the Cisco Collaboration Meeting Rooms (CMR) Hybrid Configuration Guide, available at

http://www.cisco.com/en/US/products/ps11338/products_installation_and_configuration_guides_list.html

Scheduling a CMR Hybrid meeting is a two-steps process. First, a request is sent to the WebEx Cloud to schedule the meeting on the WebEx calendar, and the WebEx Cloud responds with the meeting details that are passed to Cisco TMS. Second, Cisco TMS schedules the TelePresence meeting in its calendar. When it is the meeting start time, Cisco TMS pushes the meeting details to the TelePresence Bridge for joining the meeting on WebEx. The meeting details returned from WebEx include the date and time for the meeting, dial-in information, subject, meeting number, URL for joining the meeting, and so forth. Once the meeting has been scheduled, details for the WebEx and TelePresence portions of the meeting are sent to the host, and the host can forward the details to all participants. However, if the productivity tool is used, the meeting details are automatically included in the invitation that the host creates and sends to the meeting participants.

Single Sign On

Cisco CMR Hybrid supports scheduling the WebEx portion of the meeting in Cisco TMS using Single Sign On (SSO). This feature requires the WebEx site to have Cisco TMS provisioned as the delegated partner and to have the Partner Delegated Authentication configured. With SSO enabled in Cisco TMS, only the user's WebEx username is stored in the Cisco TMS user profile without the need of the WebEx password. When the user schedules a CMR Hybrid meeting, WebEx trusts Cisco TMS and requires only the WebEx username stored in Cisco TMS to schedule the meeting in the WebEx calendar. For Cisco TMS configuration details with SSO, refer to the Cisco Collaboration Meeting Rooms (CMR) Hybrid Configuration Guide, available at

http://www.cisco.com/en/US/products/ps11338/products_installation_and_configuration_guides_list.html

For more information regarding SSO with Cisco WebEx, refer to the white papers and technical notes available at

http://developer.cisco.com/web/webex-developer/sso-reference

Security

All communications between the enterprise network and the WebEx Cloud are encrypted (using TLS and secured RTP). Customers also have an option to turn on encryption for the SIP signal and media within the enterprise. A certificate has to be uploaded to the Cisco Expressway-E to ensure that proper handshaking takes place for the TLS connection to be functional. That certificate must be signed by a trusted Root Certificate Authority. For the list of the trusted Root Certificate Authorities, refer to the Cisco Collaboration Meeting Rooms (CMR) Hybrid Configuration Guide, available at

http://www.cisco.com/en/US/products/ps11338/products_installation_and_configuration_guides_list.html

A password is required when the TelePresence Bridge calls into WebEx to join the meeting. The password is allocated for each CMR Hybrid meeting scheduled on the WebEx calendar and is embedded in the SIP URI that is returned as part of the meeting details from the WebEx Cloud. This password is encoded into 22 bytes and qualifies for the security standards. At the start of the meeting, the TelePresence Bridge calls into WebEx using this SIP URI, and WebEx validates the password to authorize the call to join the meeting.

Deployment Options

When it is the start time for the CMR Hybrid meeting, Cisco TMS initiates the conference on the TelePresence Bridge through TelePresence Conductor for the TelePresence participants. The TelePresence Bridge makes a SIP call through TelePresence Conductor out to the WebEx Cloud using the SIP URI that was returned as part of the scheduling process and to join the conference on the WebEx side. As a result, the TelePresence Bridge establishes separate audio, active speaker video, and content sharing video streams with the cloud for the meeting. The active speaker video, content sharing video, and conference control always travels over the IP network, but the audio can travel over either the IP network or the PSTN, depending on the deployment options chosen. The various audio options available for CMR Hybrid are:

WebEx Audio Using SIP

Figure 11-28 shows the deployment of Cisco CMR Hybrid using WebEx Audio with SIP. In this option, the conference audio is established with the WebEx audio bridge through the SIP connection when the TelePresence Bridge calls out to the WebEx Cloud at the start of the meeting. The audio, active speaker video, content sharing video, and conference control are sent on the IP network from the TelePresence Bridge to the WebEx Cloud through Cisco Expressway-C and Expressway-E. As a result, the audio connection from the TelePresence Bridge cascades into the WebEx audio bridge.

WebEx Audio Using PSTN

For Cisco CMR Hybrid deployment where the in-country rule does not allow toll bypass, WebEx Audio using the PSTN could be an option. Figure 11-29 depicts this deployment. In this option, the active speaker video, content sharing video, and conference control are sent over the IP network, but the audio is established with the WebEx audio bridge through the PSTN. This option requires the deployment of a voice gateway to connect the audio call between the IP network and the PSTN. During the scheduling process, when the meeting is scheduled on the WebEx calendar, WebEx passes the dial-out number and the meeting number to Cisco TMS. At the start of the meeting, the TelePresence Bridge initiates a SIP call to the WebEx Cloud to establish the active speaker video and content sharing video. At the same time, the TelePresence Bridge dials out through the PSTN to establish an audio connection with the WebEx audio bridge. After connecting with the WebEx audio bridge, the TelePresence Bridge sends out the meeting number as a DTMF dial sequence so that WebEx can associate the audio and video call legs. As a result, the audio connection from the TelePresence Bridge cascades into the WebEx audio bridge.

Figure 11-29 Cisco CMR Hybrid Using WebEx Audio with PSTN

 

The dial-out number returned from WebEx is in full E.164 number format (for example, +14085551212). The dial plan design in Cisco Unified CM should take into account the handling of E.164 numbers. For dial plan design with Cisco Unified CM, see the chapter on Dial Plan.

Teleconferencing Service Provider Audio

The Teleconferencing Service Provider (TSP) Audio option is for customers who prefer to use the audio bridge hosted by their third-party teleconferencing service provider. The TSP Audio configuration is very similar to WebEx Audio using the PSTN configuration, except that the audio bridge is hosted by the teleconferencing service provider (see Figure 11-30). The TSP link between WebEx and TSP provides the advanced conference control features.

Figure 11-30 Cisco CMR Hybrid Using Teleconferencing Service Provider (TSP) Audio

 

During the scheduling process, in addition to the dial-out number and meeting number, extra digits for navigating through the IVR prompts on the TSP audio bridge are passed from WebEx to Cisco TMS. At the scheduled meeting start time, the TelePresence Bridge initiates a SIP call to the WebEx Cloud to establish the video connections. At the same time, the TelePresence Bridge dials out to the TSP audio bridge through the PSTN. Then the TelePresence Bridge plays out the meeting number as a DTMF dial sequence, along with additional DTMF digits to navigate through the IVR prompts on the audio bridge to start the meeting. On the WebEx side, WebEx participants start the WebEx session using the meeting client and dial into the TSP audio bridge or have callback from the audio bridge. Thus, the audio streams from TelePresence and WebEx participants are cascaded. From this point onward, information about the loudest speaker, participant list, and so forth in the WebEx side, is passed from the TSP to WebEx through the TSP link and then into the enterprise collaboration network.

The dial-out number returned from WebEx is in full E.164 number format (for example, +14085551212). The dial plan design in Cisco Unified CM should take into account the handling of E.164 numbers. For dial plan design with Cisco Unified CM, see the chapter on Dial Plan.

High Availability

There are two areas that must be considered when designing high availability for CMR Hybrid: the enterprise collaboration network and the WebEx Cloud. The WebEx Cloud is managed by Cisco and already has the redundancy built into the infrastructure. For details, see the section on Cisco WebEx Software as a Service.

In the enterprise collaboration network, utilize the clustering option from Cisco Unified CM and Cisco Expressway to provide redundancy for call control and call routing on the TelePresence endpoints. In case the primary server fails, the backup server can take over the call control and call routing functions. In addition, resiliency of the TelePresence conferencing infrastructure must be considered to handle failure of conference bridges.

For Cisco Unified CM clustering, see the chapter on Call Processing.

For Cisco Expressway clustering, refer to the latest version of the Cisco Expressway Cluster Creation and Maintenance Deployment Guide, available at

http://www.cisco.com/c/en/us/support/unified-communications/expressway-series/products-installation-and-configuration-guides-list.html

For resiliency of the TelePresence conferencing infrastructure, see the section on Cisco Collaboration Meeting Rooms Premises.

Capacity Planning

The WebEx Cloud has the built-in capability to evenly distribute the traffic and dynamically add more capacity if thresholds are exceeded. Capacity planning for Cisco CMR Hybrid involves sizing of the components running within the enterprise. The components include:

  • Call Processing Platforms

Cisco Unified CM must provide enough resources to handle the traffic generated by the TelePresence endpoints. For details, see the section on Capacity Planning for Collaboration Endpoints.

  • TelePresence Conferencing

The Cisco TelePresence Conductor, Cisco TelePresence Server, or Cisco TelePresence MCU must provide enough resources to handle the conference traffic. For details, see the section on Capacity Planning.

  • Cisco Expressway

Cisco Expressway must provide enough resources to handle the traversal call traffic for the deployment. For capacity details, see the chapter on Collaboration Solution Sizing Guidance.

Network Traffic Planning

Network traffic planning for Cisco CMR Hybrid consists of the following elements:

  • WebEx Clients Bandwidth

The WebEx meeting client uses the Scalable Video Coding (SVC) technology to send and receive video. It uses multi-layer frames to send video and it allows the receiving client to automatically select the best possible resolution to receive video. For more information regarding network traffic planning for WebEx clients, refer to the Cisco WebEx Network Bandwidth white paper available at

http://www.cisco.com/c/en/us/products/collateral/conferencing/webex-meeting-center/white_paper_c11-691351.html

  • Bandwidth from Enterprise to WebEx Cloud

For each call to the WebEx Cloud, a minimum network bandwidth of 1.1 Mbps is required between the enterprise and the WebEx Cloud. For example, if a customer is expecting five simultaneous CMR Hybrid meetings, network bandwidth of 5.5 Mbps is required. At the same time, a maximum bandwidth of 4 Mbps is supported per call.

For optimal SIP audio and video quality between the TelePresence Bridge and the WebEx Cloud, Cisco recommends setting up the video bandwidth of at least 1.3 Mbps in the region associated with each endpoint registering with Cisco Unified CM.

Design Considerations

The following design considerations apply to Cisco CMR Hybrid deployments:

  • Upgrade from previous versions of CMR Hybrid that use the Cisco TelePresence MultiPoint Switch infrastructure is not supported, and customers using those previous versions should plan for migration.
  • Every user who wants to schedule a CMR Hybrid meeting must have a host account with Cisco TelePresence Session type assigned in the WebEx site.
  • Any endpoints that can register with Cisco Unified CM and that are supported by the TelePresence Bridge can be used to join the Cisco CMR Hybrid meeting.
  • Only devices managed by the Cisco TelePresence Management Suite (TMS) can use One Button to Push (OBTP) or the Auto Connect feature to join the CMR Hybrid meeting.
  • Ensure that the Cisco Unified CM Neighbor Zone in Cisco Expressway-C is configured with Binary Floor Control Protocol (BFCP) enabled.
  • Provision Hybrid Audio in the WebEx site to allow the use of SIP audio for the TelePresence Bridge and PSTN audio for WebEx participants.
  • Cisco CMR Hybrid does not support Cisco WebEx Meetings Server.
  • The TelePresence Bridge becomes the default host if no host is present when it joins the CMR Hybrid meeting, and the host privilege is reassigned to the host when he joins using the WebEx meeting client.
  • The TelePresence Bridge will call into the WebEx Cloud at meeting start time even if no TelePresence or WebEx participant has joined yet.
  • The organizer's WebEx account and Outlook time zone should match; otherwise, the meeting scheduled in WebEx and in the Cisco TMS calendar will have different start times.
  • Enable UDP for media streaming in the firewalls for the optimal video experience.

Cisco WebEx Meeting Center Video Conferencing

Cisco WebEx Meeting Center Video Conferencing is an enterprise-grade collaboration service that provides a consistent, scalable virtual meeting room experience that combines business quality video, audio, and data sharing capabilities into a single solution delivered through Cisco WebEx Collaboration Cloud. Cisco WebEx Meeting Center Video Conferencing is included as part of the Cisco WebEx Meeting Center for user-based subscription when purchased through Annuity. It integrates with the Cisco Collaboration infrastructure and applications such as Cisco Unified CM and Cisco Expressway. Participants can join Cisco WebEx Meeting Center Video Conferencing meetings using WebEx clients, Cisco TelePresence, Cisco Jabber, or other third-party standards-based endpoints (SIP or H.323). It also provides a simple and highly secure collaboration solution from the Cisco WebEx Cloud, and participants can join the meeting regardless of their location using any device of their choice (desktop, mobile, or video endpoint). With Cisco WebEx Meeting Center Video Conferencing, users can invite others to join their personalized, always-available meeting rooms anytime, or the meeting organizer can reserve the needed rooms and resources for scheduled meetings using the productivity tools.

Architecture

Figure 11-31 illustrates the Cisco WebEx Meeting Center Video Conferencing architecture using SIP video. This architecture consists of the enterprise collaboration network and the WebEx Collaboration Cloud where all the conferencing resources are hosted, and they are connected via the Internet. The enterprise collaboration network encompasses Cisco Unified Communications Manager (Unified CM) and Cisco Expressway, and Unified CM connects with Cisco Expressway-C over a SIP trunk. Cisco Unified CM provides the call routing and call control functions for the registered video devices. Cisco Expressway provides a secure firewall traversal mechanism for calls between the enterprise and WebEx Cloud, and it routes the video calls to WebEx Cloud via the DNS zone configured inside Cisco Expressway-E. In addition, Cisco Expressway provides mobile and remote access capability to the supported Cisco video endpoints so they can register with Unified CM outside of the enterprise. In order for a participant to join the meeting and share content, the SIP device must support URI dialing and Binary Floor Control Protocol (BFCP). Without BFCP, content cannot be shared and will be seen embedded in the main video.


Note For existing Cisco VCS customers, using VCS Control as a SIP Registrar for SIP endpoints and VCS Expressway for firewall traversal is supported with the deployment.


Figure 11-31 Cisco WebEx Meeting Center Video Conferencing Architecture Using SIP Video

 

Cisco WebEx Meeting Center Video Conferencing architecture also support H.323 video devices (see Figure 11-32). In this architecture, Cisco VCS Control is the gatekeeper and provides call control for the registered H.323 endpoints. Cisco VCS Expressway provides a secure firewall traversal mechanism for calls between the enterprise and WebEx Cloud, and it routes the video calls to WebEx Cloud via the DNS zone configured inside Cisco VCS Expressway. In order for a participant to join the meeting and share content, the H.323 device must support Annex O for URI dialing and H.239 for content sharing. Without H.239, content cannot be shared and will be seen embedded in the video. In addition, H.323 devices must support either the H.245 User Input or RFC 2833 method of DTMF signaling in order to use interactive voice response (IVR) to start a meeting as a host or to join a meeting before the host.

Figure 11-32 Cisco WebEx Meeting Center Video Conferencing Architecture Using H.323 Video

 

Alternatively, Cisco WebEx Meeting Center Video Conferencing can be deployed using H.323 video without a call control system (see Figure 11-33). In this architecture, the H.323 device does not register to any gatekeeper; and when the user dials the URI, the call is routed using DNS through the firewall to the WebEx Cloud. Make sure the necessary ports on the firewall are opened so that signaling and media can pass through. For port range details, refer to the WebEx Knowledge Base article available at http://kb.webex.com/WBX264.

Figure 11-33 Cisco WebEx Meeting Center Video Conferencing Architecture Using H.323 Video Without Call Control System

 

Irrespective of SIP or H.323 devices used in the deployment, WebEx Cloud can perform the interworking between protocols. There are requirements for video devices to be used in a Cisco WebEx Meeting Center Video Conferencing deployment. For details, refer to the Cisco WebEx Meeting Center Video Conferencing Enterprise Deployment Guide, available at

http://www.cisco.com/c/en/us/support/conferencing/webex-meeting-center/products-installation-and-configuration-guides-list.html

For each participant on a video device, the audio, video, and content sharing are sent over the IP connection to WebEx Cloud, where the media are mixed with other participants, and the mixed audio, active speaker video, and content sharing are sent back to the device for display.

Cisco WebEx Meeting Center Video Conferencing uses H.264 video for active speaker and content sharing. Depending on the capability of the device and the bandwidth available, Cisco WebEx Meeting Center Video Conferencing supports active speaker video up to 720p at 30 frames per second (fps) and content video up to 720p on video devices as well as WebEx clients. WebEx meeting client has a video floor of 180p for active speaker video at the minimum bit rate of 1.2 Mbps. If the minimum bit rate cannot be maintained due to network condition (severe packets loss, for example), WebEx client will stop receiving the active speaker video but still receives content sharing as well as conference audio and sends its video to other participants. Starting with release WBS 29.11, WebEx client will periodically perform bandwidth retest and automatically reestablish active speaker video when network conditions stabilize. During the meeting, WebEx allocates the bandwidth based upon the least capable device among all WebEx clients in the conference (excluding devices running below the video floor), with a maximum bandwidth of 4 Mbps. However, if the least capable device leaves the conference, the bandwidth will be reallocated based upon the next least capable device that runs the WebEx meeting client. The allocated bandwidth determines the resolution used to display the video on the WebEx clients.

Each Cisco WebEx Meeting Center Video Conferencing session has an associated video address URI and URL. Participants dial the URI or receive callback on the video device or click on the URL to bring up the WebEx meeting client to join the meeting. A Cisco WebEx Meeting Center Video Conferencing meeting can be one of the following types:

  • Scheduled meeting

Users can use WebEx Productivity Tools (PT) to schedule Cisco WebEx Meeting Center Video Conferencing meetings. Productivity Tools is a suite of tools, including an Outlook plug-in, that allows users to schedule meetings quickly and easily within the email client. This tool suite provides seamless integration with the user's calendar, and users can schedule meetings and send the invitations to all participants directly inside the email client with a single transaction. Alternatively, user can schedule Cisco WebEx Meeting Center Video Conferencing meetings from the WebEx portal but the host has to first schedule the meeting from WebEx, and then create an invitation with meeting detail attached and send it to all the participants.

Starting with Cisco WebEx Meeting Center version WBS31R1 and using Cisco TMS 15.2 and TMSXE 5.2, WebEx Productivity Tools can be utilized to schedule Cisco WebEx Meeting Center Video Conferencing meetings with One Button to Push (OBTP). Internally, Cisco TMS creates an externally hosted conference using the SIP URI to Cisco WebEx Meeting Center Video Conferencing as the dial string. Also, Cisco TMS must have the default conference type set to OBTP.


Note Using TMS and TMSXE for Cisco WebEx Meeting Center Video Conferencing OBTP does not require integration with the TelePresence infrastructure (TelePresence Server and Conductor). On the other hand, if TMS and TMSXE are integrated with the TelePresence infrastructure, they can be used for Cisco WebEx Meeting Center Video Conferencing OBTP at the same time.


  • Permanent meeting

Meetings can be hosted in the user's personal room. Personal rooms can be enabled at the site level or per-user level in the WebEx site. When enabled, a fixed URI and URL are assigned to the user, and participants can use them to join the user's personal room. This personal room belongs to the designated user and is always on. Thus, the user can use his room for his meetings and can send an invitation to all participants with his room's URI and URL attached. With Cisco Spark Calendar Service, users can add @webex to the location field of an Outlook calendar invitation, and Calendar Connector will automatically populate the invitation with the user's personal room information. See the chapter on Mobile Collaboration, for more details.

  • Instant meeting

A user can create an instant meeting from the WebEx portal or by using the WebEx Productivity Tools, and the meeting will start immediately. Using the Meet Now configuration option, the instant meeting can be instantiated from the Meeting Center or the user's personal room or Cisco Jabber Desktop.

Security

Cisco WebEx Meeting Center Video Conferencing supports encrypted signaling and media, or a combination of encrypted and non-secure signaling and media, between the enterprise network and WebEx Cloud. For end-to-end encryption, customers can turn on encrypted signaling and media in the enterprise and use encrypted signaling and media between the enterprise network and WebEx Cloud. A certificate has to be uploaded to Cisco Expressway-E to ensure that proper handshaking takes place for encrypted signaling to be functional. That certificate can be either self-signed or signed by a trusted Root Certificate Authority (CA). For more information, refer to he Cisco WebEx Meeting Center Video Conferencing Enterprise Deployment Guide, available at

http://www.cisco.com/c/en/us/support/conferencing/webex-meeting-center/products-installation-and-configuration-guides-list.html

For SIP based calls, Cisco WebEx Meeting Center Video Conferencing supports four levels of security (in order of preference):

  • Encrypted TLS signaling with CA-signed certificates and SRTP media encryption
  • Encrypted TLS signaling with self-signed certificates and SRTP media encryption
  • Non-secure TCP signaling with SRTP media encryption
  • Non-secure TCP signaling with non-secure RTP media

Make sure to open the network ports on the firewall so that inbound and outbound traffic for signaling and media can pass through. For port range details, refer the WebEx Knowledge Base article available at http://kb.webex.com/WBX264.

All Cisco WebEx Meeting Center Video Conferencing meetings require the presence of the host to start the meeting. If the guests join before the host, they will be in the waiting room and cannot talk to each other until the host joins. In addition, a host PIN is required when the host joins the meeting from a video device.

Inside the user's personal meeting room, a Lock Room button is available that can be used to lock the room and prevent other participants from entering the user's personal room. When the room is locked and a participant tries to enter the room, that participant will be blocked until the host admits him or unlocks the room. This button is useful in case a user's personal room is used for back-to-back meetings and the host has not finished with the first meeting. The host can lock the room to prevent participants of the second meeting from entering until he finishes with the first meeting and unlocks the room.

Audio Deployment Options

For Cisco WebEx Meeting Center Video Conferencing participants using video devices, their audio, video, and content sharing are sent and received over the IP connection between WebEx Cloud and the video devices. For WebEx client participants, Cisco WebEx Meeting Center Video Conferencing supports all audio options available for the classic WebEx Meeting Center, which includes:

  • WebEx Cloud Connected Audio
  • WebEx Audio using VoIP
  • WebEx Audio using PSTN
  • Teleconferencing service provider audio

High Availability

In the enterprise collaboration network, utilize the clustering option with Cisco Unified CM and Cisco Expressway to provide redundancy for call control with video devices and firewall traversal calls. If the primary server fails, the backup server can take over the call control and call handling functions.

For Cisco Unified CM clustering, see the chapter on Call Processing.

For Cisco Expressway clustering, refer to the latest version of the Cisco Expressway Cluster Creation and Maintenance Deployment Guide, available at

http://www.cisco.com/c/en/us/support/unified-communications/expressway-series/products-installation-and-configuration-guides-list.html

Capacity Planning

Cisco WebEx Meeting Center Video Conferencing meetings support up to 25 standards-based video devices, 500 WebEx participants with video enabled, and 500 WebEx participants with audio only.


Note Each screen in a multi-screen video device counts as one video device. For example, if a triple-screen immersive system joins the Cisco WebEx Meeting Center Video Conferencing meeting, it consumes 3 video devices from the video device capacity limit.


Capacity planning for Cisco WebEx Meeting Center Video Conferencing involves sizing of the components running within the enterprise. The components could include:

  • Cisco Unified CM

Ensure that Unified CM has enough resources and capacity to handle the traffic generated by the video endpoints and IP phones for Cisco WebEx Meeting Center Video Conferencing meetings. For capacity details, see the chapter on Collaboration Solution Sizing Guidance.

  • Cisco Expressway

Cisco Expressway must provide enough resources to handle the traversal call traffic for the deployment. For capacity details, see the chapter on Collaboration Solution Sizing Guidance.

Network Traffic Planning

Network traffic planning for Cisco WebEx Meeting Center Video Conferencing consists of the following elements:

  • WebEx Clients bandwidth

The WebEx meeting client uses the Scalable Video Coding (SVC) technology to send and receive video. It uses multi-layer frames to send video, and the receiving client automatically selects the best possible resolution to receive video that typically requires 1.2 to 3 Mbps available bandwidth. For more information regarding network traffic planning for WebEx clients, refer to the Cisco WebEx Network Bandwidth white paper, available at

http://www.cisco.com/c/en/us/products/collateral/conferencing/webex-meeting-center/white_paper_c11-691351.html

  • Bandwidth for video device from enterprise to WebEx Cloud

For optimal SIP audio and video quality, Cisco recommends setting up the video bandwidth for at least 1.5 Mbps per device screen in the region associated with the endpoint registering with Cisco Unified CM. For example, if a triple-screen device registers with Unified CM, video bandwidth of 4.5 Mbps should be allocated in the associated region.

Design Considerations

Consider the following recommendations when deploying Cisco WebEx Meeting Center Video Conferencing:

  • Enable UDP for media streaming in the firewalls for the optimal video experience.
  • Open network ports on firewalls to allow inbound and outbound signaling and media traffic. For details, refer to the he Cisco WebEx Meeting Center Video Conferencing Enterprise Deployment Guide, available at

http://www.cisco.com/c/en/us/support/conferencing/webex-meeting-center/products-installation-and-configuration-guides-list.html

  • Ensure that Binary Floor Control Protocol (BFCP) is enabled in the Unified CM Neighbor Zone in Cisco Expressway-C and that BFCP is also enabled in the SIP profile associated with the SIP trunk between Unified CM and Expressway-C.
  • For information on the video devices tested with Cisco WebEx Meeting Center Video Conferencing, refer to the WebEx Meeting Center Video Compatibility and Support document, available at

https://help.webex.com/docs/DOC-6428