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Cisco Unified Communications SRND Based on Cisco Unified CallManager 4.x
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Preface
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Introduction
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IP Telephony Deployment Models
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Network Infrastructure
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Gateways
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Cisco Unified CallManager Trunks
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Media Resources
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Music on Hold
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Call Processing
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Call Admission Control
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Dial Plan
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Emergency Services
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Third-Party Voicemail Design
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Cisco Unity
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Cisco Unity Express
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Cisco Unified MeetingPlace Integration
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Cisco Unified MeetingPlace Express
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IP Video Telephony
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LDAP Directory Integration
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IP Telephony Migration Options
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Voice Security
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IP Telephony Endpoints
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Device Mobility
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Recommended Hardware and Software Combinations
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Glossary
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Index
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Table Of Contents
Cisco Unified MeetingPlace Integration
MeetingPlace Server Recommendations
MeetingPlace H.323/SIP IP Gateway
Sizing a MeetingPlace Deployment
Sizing Voice Conferencing Usage
Video Integration and Sizing the MCU
MeetingPlace Network Infrastructure
Connectivity Between MeetingPlace and IP Telephony Components
Call Admission Control and Bandwidth Provisioning
Rate and Codec Selection for Audio and Video
MeetingPlace Web Session Network Utilization
Demilitarized Zone (DMZ) Requirements
Internet Protocol (IP) Network
Protocols Supported by the MeetingPlace Audio Server
Protocols Supported by Other MeetingPlace Components
Public Switched Telephone Network (PSTN)
Dialing into an Audio-Only Conference Using Video Endpoints
Enhanced Media Processor (EMP) Requirement
Dynamic H.323 Addressing in Cisco Unified CallManager
Cisco Unified CallManager Redundancy Groups and H.323 Clients
Reservationless Single Number Access (RSNA)
MeetingPlace H.323/SIP IP Gateway
Cisco Unified MeetingPlace Integration
Last revised on: February 13, 2008
This chapter describes the technical and design issues for incorporating Cisco Unified MeetingPlace into an existing Cisco Unified Communications network. The fundamental network infrastructure and IP Telephony design considerations for MeetingPlace remain the same as the IP Telephony design considerations for Cisco Unified CallManager, and this chapter assumes that you have basic knowledge and experience with Cisco Unified Communications.
This chapter focuses mainly on the integration and design issues for combining both Cisco Unified MeetingPlace and IP Telephony in one converged network. The time-division multiplexing (TDM) connection from MeetingPlace to the public switched telephone network (PSTN) is not a consideration for this setup because the PSTN access is typically provided through Cisco Unified CallManager by means of a voice gateway.
This chapter does not cover other MeetingPlace components that do not affect the integration, such as the MeetingPlace components for Outlook, Lotus Notes, Email (Simple Mail Transfer Protocol, or SMTP), Directory Services, and Instant Messaging. Also, specific product-level information about MeetingPlace (such as feature descriptions and configuration options) is beyond the scope of this document. For more information about MeetingPlace, refer to the product documentation available at http://www.cisco.com.
MeetingPlace Server Recommendations
Table 15-1 lists the recommended Cisco Media Convergence Server (MCS) models to use in various deployment scenarios.
The following guidelines also apply to MCS deployments:
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Deployments with up to 50 Web Conferencing user licenses can run web conferencing on the same MCS with the bundled software and other options.
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Deployment Models
This section covers the major deployment models used to deploy MeetingPlace with Cisco Unified CallManager. For the purpose of this section, assume that the MeetingPlace system is placed at a major site that includes a Cisco Unified CallManager cluster.
Figure 15-1 shows Cisco Unified MeetingPlace 5.3 integrated with a comprehensive Cisco IP Communications topology. The Cisco Unified CallManager cluster (running Cisco Unified CallManager 4.0 or above) includes a video deployment (both SCCP and H.323 video endpoints) with the video conferencing capability provided by a Multipoint Control Unit (MCU). The H.323-to-H.320 video gateway handles video calls to the PSTN. Through the MeetingPlace H.323/SIP IP Gateway, Cisco Unified CallManager fully utilizes the rich-media features provided by MeetingPlace with its audio, video, and web conferencing solution, which external users can also access from the Internet.
Figure 15-1 Example Topology for Integrating MeetingPlace with IP Telephony
The following sections describe the major deployment models for integrating MeetingPlace with Cisco Unified CallManager.
Single Site
The single-site model for IP Telephony and MeetingPlace consists of a call processing agent located at a single site and a LAN or metropolitan area network (MAN) to carry voice, video, and collaboration traffic throughout the site. Conference participants beyond the LAN or MAN use the public switched telephone network (PSTN). If an IP WAN is incorporated into the single-site model, it is for data and web collaboration traffic only; no telephony services are provided over the WAN.
For a more detailed explanation of the IP Telephony single-site deployment model, see the chapter on IP Telephony Deployment Models, page 2-1.
MeetingPlace should be deployed within the data center to provide maximum resilience for the conferencing and collaboration system. Cisco recommends using Media Gateway Control Protocol (MGCP) voice gateways in a single-site model; however, an H.323/H.320 Cisco Unified Videoconferencing Gateway is required to support external video participants. The recommended method for providing external access is to use either a toll-free service or dedicated direct inward dial (DID) numbers for the MeetingPlace pilot numbers via Cisco Unified CallManager gateways, but you also have the option to use dedicated PSTN T1 or E1 lines directly connected to MeetingPlace. Typical systems include a single toll-free number, a single DID or central office (CO) number, and an internal dialing number provided for users to dial into an audio conference session. Unique DID numbers can be used for dedicated crisis management or unique meeting IDs that are always available (24/7/365) for other applications.
Multisite WAN with Centralized Call Processing
The multisite WAN model with centralized call processing consists of a single call processing agent that provides services for many sites and uses the IP WAN to transport voice and /or video traffic between the sites. The IP WAN also carries call control signaling between the central site and the remote sites.
For a more detailed explanation of the IP Telephony multisite deployment model with centralized call processing, see the chapter on IP Telephony Deployment Models, page 2-1.
MeetingPlace should be deployed at the main site to provide centralized conferencing and collaboration services for all sites. When considering the number of conference participants from each site, plan for the required WAN bandwidth and/or the number of PSTN calls to the central-site MeetingPlace. If there is insufficient bandwidth on the WAN, users will have to redial via the PSTN using either a toll-free service or a dedicated CO or DID number. If remote sites enter Survivable Remote Site Telephony (SRST) mode or switch to Cisco Unified CallManager Express for connectivity loss to the central Cisco Unified CallManager cluster, then the only access to MeetingPlace conferencing is via the PSTN, and video support is not automatically included. The web collaboration traffic, however, could still use a backup data path if one is available.
Multisite WAN with Distributed Call Processing
The multisite WAN model with distributed call processing consists of multiple independent sites, each with its own call processing agent connected to an IP WAN that carries voice and video traffic between the distributed sites.
For a more detailed explanation of the IP Telephony multisite deployment model with distributed call processing, see the chapter on IP Telephony Deployment Models, page 2-1.
MeetingPlace may be deployed at one, some, or all of the distributed call processing sites. Access to the various MeetingPlace systems in different Cisco Unified CallManager clusters is by means of intercluster trunks and/or the PSTN. Web collaboration is always via the IP WAN. For maximum flexibility, implement a uniform dial plan to ensure that MeetingPlace is accessible by any user on any Cisco Unified CallManager cluster using a MeetingPlace connected to the same or any other Cisco Unified CallManager cluster. MeetingPlace also provides a feature called multiserver meetings, which are cascaded conferences that can conserve bandwidth or PSTN calls when multiple users at multiple sites are in the same conference. (For more information on multiserver meetings, see the chapter on Conferencing.)
Clustering Over the IP WAN
This model deploys a single Cisco Unified CallManager cluster across multiple sites that are connected by an IP WAN with QoS features enabled.
For a more detailed explanation of clustering over the WAN, see the chapter on IP Telephony Deployment Models, page 2-1.
With clustering over the WAN, you can deploy multiple MeetingPlace Audio Servers at different Cisco Unified CallManager sites. (See the information on Dual Conference Servers in the section on Disaster Recovery.) The user profile information can be synchronized between the servers by using the MeetingPlace Directory Service Integration option.
On Cisco Unified CallManager, you can configure route groups and route lists so that, if one of the MeetingPlace Audio Servers fails, the calls will still be routed to another MeetingPlace system. The meeting information does not transfer between servers, so the administrator must manually upload the meeting information from the failed system onto another system for use.
If there is a WAN outage, each server is still able to operate and serve its local users. For remote users to join the meeting on the same audio server, you can configure route groups and route lists on Cisco Unified CallManager to reroute the calls.
With multiple audio servers, Cisco highly recommends turning off the Vanity ID feature (which allows users to choose and set their own meeting IDs) in order to support failover mode and reduce the probability that meeting IDs would conflict during uploads.
MeetingPlace Components
This section describes the following major components that affect the design of a MeetingPlace system:
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MeetingPlace Audio Server
The MeetingPlace Audio Server (8112 or 8016) provides digital signal processor (DSP) resources for audio conferencing capability. Additionally, the MeetingPlace server acts as a scheduling module for web, audio, and video conferences. Based on the reservation, the MeetingPlace server will control how many users can join the conference as well as the duration of the meeting.
MeetingPlace H.323/SIP IP Gateway
The MeetingPlace Audio Server was originally designed for TDM connectivity, via T1/E1 trunks, either to a service provider or behind a PBX. To incorporate it into an existing Cisco Unified Communications network, the MeetingPlace H.323/SIP IP Gateway is required to link the two networks. Although it is called a gateway, it is not a hardware gateway but a software application that resides on an MCS server and is used to communicate between the MeetingPlace Audio Server and Cisco Unified CallManager. (See Figure 15-2.)
Figure 15-2 MeetingPlace H.323/SIP IP Gateway Used to Interface
The MeetingPlace H.323/SIP IP Gateway supports H.323 and SIP simultaneously. For details on the protocols supported by the MeetingPlace H.323/SIP IP Gateway, refer to the section on Interoperability Protocols.
MeetingPlace audio servers can support mixed TDM and IP connections. The capacities for a mixed environment depend on the combination of both TDM ports and IP cards (user licenses) in the system.
Although the MeetingPlace H.323/SIP IP Gateway can coexist on the same Cisco Media Convergence Server (MCS) with other MeetingPlace applications (such as MeetingPlace Web, SMTP Email, Video Integration, and so forth), in a pure IP setup Cisco recommends that you install the MeetingPlace H.323/SIP IP Gateway independently on a single MCS so that audio communications will not be affected if any problems occur with the web or video portions. However, for small deployments the MeetingPlace H.323/SIP IP Gateway can coexist with the MeetingPlace Web Conferencing Service, and the performance and capacity of the web conference should not be impacted significantly because it does not use much of the MCS CPU resources.
A single MeetingPlace Audio Server can be connected to multiple MeetingPlace H.323/SIP IP Gateways. If one MeetingPlace H.323/SIP IP Gateway fails, calls in progress are dropped, and new calls are routed to the secondary MeetingPlace H.323/SIP IP Gateway. For outdialing from the MeetingPlace Audio Server, the server chooses the MeetingPlace H.323/SIP IP Gateway with the least activity. For more information, refer to the section on Redundancy and Load Balancing.
Each MeetingPlace H.323/SIP IP Gateway can support only one Cisco Unified CallManager without using a gatekeeper, and multiple Cisco Unified CallManagers with a gatekeeper. Each MeetingPlace Audio Server can connect to a maximum of 16 MeetingPlace H.323/SIP IP Gateways for the purposes of redundancy, load balancing, and support for multiple Cisco Unified CallManager clusters.
An Audio Server can support up to a maximum of 16 GWSIM MCS servers with MeetingPlace applications installed, so multiple IP gateways can be supported (until the combined MCS application server count = 16).
Note
The MeetingPlace H.323/SIP IP Gateway can communicate with Cisco Unified CallManager in one or more of the following ways:
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To implement this method, configure the connection to the MeetingPlace H.323/SIP IP Gateway as a SIP Trunk in Cisco Unified CallManager.
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To implement this method, configure the connection to the MeetingPlace H.323/SIP IP Gateway as an H.323 gateway in Cisco Unified CallManager.
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In this method, Cisco Unified CallManager and the MeetingPlace H.323/SIP IP Gateway register with a gatekeeper, and the gatekeeper handles the call routing.
Sizing a MeetingPlace Deployment
Sizing a Cisco Unified MeetingPlace deployment involves the following major considerations:
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When sizing a MeetingPlace system, always start with the 8100 Audio Server, and use the most accurate estimate available for the average conferencing minutes per month. This estimate can be derived from current billing information obtained from the conferencing service provider(s). For example, if the billing information contains a yearly total conferencing minutes, then you can either divide the total by 12 or select a peak month to use as the estimate. User surveys and feedback can also be helpful in deriving the average monthly usage.
In addition, you should add some growth factor (typically 10% to 30%) for at least the first year and possibly subsequent years. If you are incorporating the Cisco Unified MeetingPlace Outlook Integration option (which enables end users to easily schedule, notify, and attend voice, web, or video meetings), then your growth factor might be larger (possibly 30% to 50%) based on projected usage.
Use the following general guidelines when sizing a MeetingPlace solution:
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You can also use these assumptions to compare to the billing data.
Sizing Voice Conferencing Usage
The general equation for calculating the number of required voice user licenses is:
Number of MeetingPlace voice user licenses =
(Average conferencing minutes per month + growth factor) / Baseline
Round any fractions up to the next whole number.
Table 15-2 lists the baseline values to use in this equation.
For example, if you estimate the average monthly usage of your system to be 528,000 conferencing minutes, the required number of user licenses would be:
Number of MeetingPlace voice user licenses = 528,000 * (1 + 20%) / 2500 = 254
Sizing Web Conferencing Usage
You can calculate the number of required web user licenses based on either a ratio of the total voice user licenses or service provider billing information about web conference usage. Typical deployments use 25% to 50% as many web conferencing licenses as voice conferencing licenses (25% is most common), but some enterprises deploy equal number of voice and web user licenses.
For example, a deployment with 240 voice user licenses would need 60 web user licenses to provide 25% coverage for web users. One MCS 7845 could handle the web services for this system, while also providing for growth up to 200 web user licenses.
Web conferencing directly affects the number of Cisco MCS 7800 Series servers that are required. An MCS 7835 co-located with other MeetingPlace integration modules can accommodate approximately 50 web sessions, while an MCS 7845 can accommodate up to approximately 200 web sessions (depending on the actual type of usage at a particular peak time). Therefore, sizing the web conferencing resources is critical to the overall design and placement of the solution in the enterprise data environment.
Web conferencing usage typically experiences faster growth rates than voice conferencing usage, and it depends on the amount of web collaboration used within the enterprise. For web conferencing, try to estimate growth for the next two to five years (typically 20% to 30% growth per year).
Video Integration and Sizing the MCU
MeetingPlace Video Integration is a system-wide software module, and no user licenses are required for video conferencing. The number of supported video users depends on the number of H.323 ports available on the MCU for MeetingPlace to control. The IP/VC 3511 MCU provides 15 ports (either H.323 or SCCP but not both), and the 3540 MCU provides 30, 60, or 100 ports (either H.323, SCCP, or a mixture of both). (See Table 15-3.)
MeetingPlace Video Integration allows users to schedule video conferences with only a single IP/VC MCU. MeetingPlace currently does not support multiple MCUs or cascaded video meetings. You may deploy other third-party MCUs to support other videoconferencing applications that do not integrate with the MeetingPlace Video Integration solution.
MeetingPlace Video can reside on only one MeetingPlace Web MCS in the system, which can be either an internal or external web server (but not both). Video meetings are enabled only on the web server where MeetingPlace Video is installed, and they are not supported on any other web server in the system. Voice and web conferences are still supported on all servers. The MeetingPlace Video Integration software can reside on the same MCS as the Web Conferencing module, along with other MeetingPlace integration modules (such as Outlook, Notes, Directory Services, and so forth), depending on the expected number of web user licenses and the total number of web servers.
Because it can reside on only one web server, the MeetingPlace Video solution does not currently provide redundancy.
You can use the same Cisco Unified Videoconferencing 3540 or 3511 MCU for MeetingPlace video integration with other SCCP video telephony endpoints, and a single IP/VC 3540 MCU can support both ad-hoc video telephony calls (SCCP controlled) and MeetingPlace H.323 conferences. In such a deployment, you allocate some of the ports on the MCU to support SCCP for video telephony and the rest of the ports to support H.323 for MeetingPlace. This dual mode is not supported on the IP/VC 3511 MCU. To support both modes with the IP/VC 3511 MCU, you would have to deploy one 3511 MCU for video telephony and a separate 3511 MCU for MeetingPlace Video Integration. (See Table 15-3.)
Table 15-3 Possible MCU Port Allocations for SCCP and H.323
100%
0%
100%
0%
100%
0%
100%
0%
0%
100%
50%
50%
75%
25%
70%
30%
0%
100%
50%
50%
50%
50%
25%
75%
30%
70%
0%
100%
0%
100%
1 This MCU supports 16 ports for SCCP at 128 or 384 kbps.
MeetingPlace Network Infrastructure
This section explains the requirements of the network infrastructure needed to build a MeetingPlace conferencing solution in an existing Cisco Unified Communications enterprise environment. The requirements in this section must be used in conjunction with the network infrastructure requirements described in the chapter on Network Infrastructure, page 3-1.
Figure 15-3 illustrates a typical network configuration for a MeetingPlace conferencing solution.
Figure 15-3 Typical MeetingPlace Conferencing Solution in an Enterprise IP Telephony Network
The following sections list the main requirements for the network that connects the MeetingPlace Audio Server and its components to the IP Telephony infrastructure:
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Connectivity Between MeetingPlace and IP Telephony Components
Adhere to the following practices when connecting MeetingPlace to the IP Telephony network:
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Quality of Service (QoS)
This section discusses the following QoS mechanisms as they relate to MeetingPlace:
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Traffic Classification
Traffic classification is a keystone for voice quality in congested networks. Voice packets must be classified or differentiated from data packets and must be handled with high priority. Some of the voice traffic is marked at its source, but other traffic needs to be classified as close as possible to the entry points of the network.
The MeetingPlace Audio Server has the following traffic classification characteristics:
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The MeetingPlace Audio Server allows one of the following Layer 3 settings to be applied to RTP packets:
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Note
There is no mechanism within the MeetingPlace Audio Server to mark the voice signaling traffic at its source. The traffic between all MeetingPlace components must be marked as soon as it enters the network. The traffic between MeetingPlace components is usually from the Gateway System Integrity Manager (GWSIM) application, which is loaded on an MCS 8100 Series server and is always present with the MeetingPlace integration software modules (Web, Outlook, Notes, Video, and so forth). When this traffic traverses the WAN, it must be given the same priority as the other voice signaling traffic.
Cisco recommends that you use packet markings that are consistent with other devices. Refer to the chapter on Network Infrastructure, page 3-1, for detailed information.
Call Admission Control and Bandwidth Provisioning
The MeetingPlace Audio Server and other MeetingPlace components do not have any call admission control mechanisms of their own. The MeetingPlace Audio Server can accept IP calls or conference requests until all ports or resources have been exhausted. This behavior can result in degraded voice quality for all audio streams across a given link if sufficient bandwidth is not available.
Call admission control should be implemented in deployments involving bandwidth-restricted links such as WAN links. Refer to the chapter on Network Infrastructure, page 3-1, for detailed information regarding call admission control.
Rate and Codec Selection for Audio and Video
Audio
In an audio- only conference, the audio codec selected for a voice call depends on the regions configuration in Cisco Unified CallManager. The codec specified in the regions configuration in Cisco Unified CallManager must match the audio codec for which the MeetingPlace Audio Server is configured. Most commonly used codecs are G.711 A-law or mu-law and G.729. MeetingPlace supports both types of codecs, but only G.711 is enabled by default. If G.729 support is also desired, you should enable it during Audio Server initialization.
Video
Cisco Unified CallManager Release 5.0 supports H.261, H,263, and H.264 codecs. For a video conference, the video rate can be configured in the following places:
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The maximum video rate in a MeetingPlace user profile is set to 384 kbps. The video rate specified in the Cisco Unified CallManager region configuration will take effect only if the bandwidth requested or the video rate configured in the MCU or MeetingPlace user profile exceeds the region configuration.
For example, consider the following video bandwidth settings:
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In this case, if a video user dials into the MCU to join the conference, the video rate selected will be 384 kbps which corresponds to the setting on the MCU service view. The video bandwidth setting in the MeetingPlace user profile will not take effect at all because MeetingPlace Video is not involved in the call flow.
In the case of outdialing to a user, the MeetingPlace Video application will pass the video bandwidth value (256 kbps) in the MeetingPlace user profile for this user to the MCU. The MCU compares this value with its own service view value (384 kbps) and selects the lowest bandwidth value (in this case, 256 kbps) for the video conference and extends a call request to the Cisco Unified CallManager video telephony endpoint or the H.323 video endpoint. The video bandwidth selected for the call depends on the video endpoint. In the case of a Skinny Client Control Protocol (SCCP) video endpoint, the bandwidth is selected according to the video rate settings on the MCU service view and the Cisco Unified CallManager region. (In this case 384 kbps is selected.) However, in the case of an H.323 video endpoint, the bandwidth selection is based on all three video rate settings. (In this case, 256 kbps is selected.)
Figure 15-4 shows the algorithm for selecting the video bandwidth.
Figure 15-4 Algorithm for Selecting the Video Bandwidth
The Cisco Unified CallManager region bandwidth setting (512 kbps in our example) takes effect only when the MCU or MeetingPlace Video application requests more bandwidth than the region setting (in this case, more than 512 kbps) and the endpoint dials into the video meeting. If the endpoint user employs the outdial feature to connect, then the MeetingPlace profile bandwidth setting is used.
MeetingPlace Web Session Network Utilization
MeetingPlace Web sessions generate the following amounts of network traffic:
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Cisco Unified MeetingPlace Web Conferencing will transmit 24-bit color if the bandwidth is available and will automatically reduce it to 8-bit color for slower connections. This determination is made automatically by attempting to transmit at the higher rate and reducing that rate if data is significantly delayed or lost.
Jitter
Variations in the packet delay, known as jitter, can seriously affect the voice quality. A jitter buffer smooths out variations in network delay. The MeetingPlace Audio Server contains the following parameters for setting the jitter buffer:
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In most cases, you should not change the default values of these parameters. These values may be adjusted if voice quality issues are encountered.
Domain Name System (DNS)
The MeetingPlace Audio Server does not support DNS internally. Cisco recommends assigning static IP addresses to all MeetingPlace Audio Server components and MCS components in your implementation. The Audio Server attempts to use IP addresses when accessing other servers. MeetingPlace components (such as MeetingTime System Administration client) may use host names to contact the MeetingPlace Audio Server. MeetingPlace components may also use host names to communicate with other Cisco Unified Communications products.
Network Time Protocol (NTP)
The MeetingPlace Audio Server uses NTP to synchronize its clock with the network time server. The MeetingPlace Audio Server may be configured with up to three NTP servers. MeetingPlace windows-based components can use the w32time.exe service to act as the NTP client or server. Cisco highly recommends that you use only one NTP server across the entire enterprise network so that the clocks are synchronized on all components.
Demilitarized Zone (DMZ) Requirements
MeetingPlace components placed in a DMZ allow external users access to meetings. For external users to access web conferences, a separate MeetingPlace Web server can be placed in the DMZ. Internal users can still have confidential meetings using an internal MeetingPlace Web server.
In a DMZ deployment, internal users have full web access and MeetingPlace capabilities (schedule, find, attend, access attachments, and record), while external users are allowed only to attend meetings. When the meeting starts, the internal users are redirected to the external web server.
Figure 15-5 illustrates this type of deployment.
Figure 15-5 MeetingPlace Web Conferencing with a DMZ for External Users
If MeetingPlace components in a DMZ are separated from internal MeetingPlace components by security policies such as a firewall, it will be necessary to open certain Transmission Control Protocol (TCP) or User Datagram Protocol (UDP) ports through the firewall for integration to work properly.
No ports need to be opened from the DMZ to the internal network, but the following ports from the internal network to the DMZ are needed:
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The following ports are open between the DMZ and the Internet:
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The DNS service must be split (one internal DNS server and one external DNS server) for the following reasons:
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The internal DNS server resolves queries from inside the corporate network only, while the external DNS server holds the publicly addressable entities for the corporation's domain. The DNS servers must map both URLs and IP addresses.
Note
Table 15-4 lists all the TCP and UDP ports used by the MeetingPlace Audio Server and its components.
Interoperability Protocols
This section describes the communication protocols used by the MeetingPlace Audio Server and its components to achieve interoperability with other Cisco products. Interoperability with the Cisco Unified MeetingPlace conferencing solution involves integration with up to two networks:
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The MeetingPlace components (MeetingPlace H.323/SIP IP Gateway, MeetingPlace Web server, and MeetingPlace Video application) integrate directly with other products, and the MeetingPlace components then communicate with the MeetingPlace Audio Server.
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The MeetingPlace Audio Server integrates directly with other products through a TDM interface on the Audio Server.
Internet Protocol (IP) Network
This section describes the protocols used to integrate MeetingPlace components with an IP network. The protocols are of the following types:
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Protocols Supported by the MeetingPlace Audio Server
The MeetingPlace Audio Server uses the following protocols or applications to integrate with an IP Communications system:
GWSIM
The Gateway System Integrity Manager (GWSIM) is a proprietary application that serves as an interface between the MeetingPlace Audio Server and its components. The Cisco Unified MeetingPlace GWSIM is based on a client/server architecture, which provides a means of exchanging information with the remote MeetingPlace components. The MeetingPlace Audio Server manages all the meeting information using a System Integrity Manager (SIM). The SIM uses the IP network to exchange all signaling and conferencing information with the GWSIM agent installed on the MeetingPlace components. This communication uses a proprietary Cisco protocol using TCP ports 5001 and 5003, and it is commonly referred to as GWSIM communication.
Real-Time Transport Protocol (RTP)
The MeetingPlace Audio Server uses User Datagram Protocol (UDP) for transmission of real-time audio directly to the endpoints (such as IP phones, IP/VC MCU, and so forth). The Audio Server, by default, is enabled to use only the G.711 codec for RTP. However, you can change this default to the desired codec by using the command line interface.
The MeetingPlace Audio Server requires use of the MeetingPlace H.323/SIP IP Gateway in order to support H.323 and SIP.
Protocols Supported by Other MeetingPlace Components
This section describes the protocols used by MeetingPlace components, other than the MeetingPlace Audio Server, to integrate with IP Communications networks.
H.323
H.323 networks can be integrated with MeetingPlace for a feature-rich conferencing solution. As mentioned earlier, because the MeetingPlace Audio Server does not natively support H.323, the MeetingPlace H.323/SIP IP Gateway is required to provide H.323 protocol support. The Cisco Unified MeetingPlace H.323/SIP IP Gateway supports H.323 and communicates directly with the other H.323 elements in the network to enable conferencing with the Audio Server.
The MeetingPlace H.323/SIP IP Gateway integrates directly with Cisco Unified CallManager or Cisco Unified CallManager Express using H.323. The number of simultaneous H.323 calls that a MeetingPlace H.323/SIP IP Gateway can handle depends on the maximum IP port limits of the Audio Server. Optionally, a gatekeeper can also be used for address resolution and bandwidth control when integrating with Cisco Unified CallManager or CallManager Express.
The MeetingPlace H.323/SIP IP Gateway uses the following TCP and UDP ports for the indicated purposes:
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Note
Figure 15-6 shows the interoperability of MeetingPlace with Cisco Unified CallManager and CallManager Express.
Figure 15-6 Integration with Cisco Unified CallManager and CallManager Express Using H.323
SIP
The MeetingPlace H.323/SIP IP Gateway also provides the ability to interconnect Session Initiation Protocol (SIP) networks to the Cisco Unified MeetingPlace Audio Server. The MeetingPlace H.323/SIP IP Gateway integrates directly with a SIP proxy server or SIP gateway and converts the SIP messages to GWSIM messages, thus allowing SIP endpoints to use the MeetingPlace Audio Server for conferencing.
Cisco Unified CallManager Release 4.2 can be integrated directly with the MeetingPlace H.323/SIP IP Gateway using a SIP trunk on Cisco Unified CallManager. Using a SIP trunk between Cisco Unified CallManager and the MeetingPlace H.323/SIP IP Gateway has three significant requirements:
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The MeetingPlace H.323/SIP IP Gateway uses the following UDP ports for SIP:
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Figure 15-7 shows the interoperability of MeetingPlace with Cisco Unified CallManager.
Figure 15-7 SIP Integration
Note
XML Application
The MeetingPlace Audio Server can handle only audio communications. To support video conferencing, it requires an external MeetingPlace Video application that integrates directly with the IP/VC Multipoint Control Unit (MCU). Cisco Unified MeetingPlace Video Integration communicates with the Cisco Unified Videoconferencing MCU using XML messaging through TCP port 3336.
Figure 15-8 shows the interoperability of MeetingPlace with the IP/VC MCU:
Figure 15-8 Video Integration
Public Switched Telephone Network (PSTN)
The Cisco Unified MeetingPlace Audio Server connects to the PSTN either directly or through a PBX using digital (T1 or E1) or analog trunks. The MeetingPlace Audio Server is also reachable by PSTN via Cisco Unified CallManager with a voice gateway connected.
Digital Trunks
Cisco Unified MeetingPlace supports T1 and E1 digital trunks, as described in the following sections.
T1
Cisco Unified MeetingPlace supports the following protocols for T1 digital trunks:
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T1 Smart Blades support T1 directly connected from either the PSTN or PBXs. The multi-access blades (MA-4 or MA-16) support T1-PRI or E1-PRI digital connections to a PBX or to the PSTN. The framing for the digital lines can be either Extended Superframe (ESF) or D4 framing. The digital lines can use either Binary 8-Zero Substitution (B8ZS) or Alternate Mark Inversion (AMI) coding.
Note
Digital connections usually provide either E&M or ground start (GST) signaling. On T1 circuits configured for E&M signaling, MeetingPlace can receive only dialed number information - either Direct Inward Dial (DID/DDI) or Dialed Number Identification Service (DNIS). MeetingPlace can use dialed number information to connect the caller directly to a meeting or to determine the MeetingPlace services to which the caller has access.
Cisco Unified MeetingPlace also supports fractional T1 services.
E1
Cisco Unified MeetingPlace supports the following protocols for E1 digital trunks:
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Note
Each E1 protocol allows software configuration of signaling options. The signaling options must match with the options configured on the PBX or PSTN switch.
Note
Table 15-6 Port Capacities for a Mixed System with T1-PRI and IP Ports
Number of IP ports
0
120
400
480
960
Maximum number of PRI ports
736
368
368
368
0
Total ports
736
488
768
848
960
Table 15-7 Port Capacities for a Mixed System with E1 and IP Ports
Number of IP ports
0
120
384
480
960
Maximum number of E1 ports
960
480
480
480
0
Total ports
960
600
864
960
960
For more information, refer to the Getting Started Guide for Cisco Unified MeetingPlace Audio Server and the Configuration Guide for Cisco Unified MeetingPlace Audio Server, both available at
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/tsd_products_support_series_home.html
Conferencing
Cisco Unified MeetingPlace supports the following types of conferencing:
Audio Conferencing
There are two platforms for the MeetingPlace Audio Server:
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The maximum number of simultaneous meetings equals the number of ports divided by two. The servers support up to 550 sessions (participants) per meeting. However, you can cascade up to three Audio Servers to achieve a total of 1650 audio sessions in a single meeting.
Call Flow
The media request (signaling) is handled by the MeetingPlace H.323/SIP IP Gateway on behalf of the MeetingPlace Audio Server, while the media stream flows directly between the MeetingPlace Audio Server and IP phones. Figure 15-9 illustrates the call flow for inbound calls, and Figure 15-10 illustrates the call flow for outbound calls.
Figure 15-9 Audio Dial-in Call Flow for MeetingPlace Integrated with Cisco Unified CallManager
Figure 15-10 Audio Outdial Call Flow from Web Interface
Meeting Type
MeetingPlace supports two main types of audio meetings:
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While scheduling this type of meeting, you can specify the meeting parameters, such as meeting ID, number of participants, and so forth. When participants join, they are connected directly to the main meeting. Immediate meetings are basically scheduled meetings in terms of the resource reservation, with the only difference being the start time (starting immediately rather than in the future).
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Any profile user can start a reservationless meeting from the phone if that feature is enabled. The user must sign in with a profile ID and password, which starts the meeting. The database also stores information about who started the meeting.
Reservationless mode is both a system-wide parameter and a user group parameter. It can be turned on for the whole system and off for a particular set of end users, but not the reverse. This parameter will change the prompts that play when participants join the meetings.
Both types of meetings can reside on the same MeetingPlace Audio Server, and they share the same pool of conference ports. Enabling the reservationless mode does not affect the capability to schedule the standard type of meetings simultaneously. However, reservationless meetings automatically set the user's profile number as the meeting ID. Therefore, if the reservationless mode is enabled, user profile numbers become reserved and cannot be assigned manually as meeting IDs for standard scheduled meetings.
Port Management
The MeetingPlace Audio Server provides the following types of ports:
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These ports are reserved for scheduling, attending, and listening to recorded meetings.
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These ports are the ones in use or reserved for meetings. Conference ports also include the following types of special-usage ports:
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These ports do not exist physically. They are software ports configured to allow users to schedule more ports than are actually available, based on the assumption that there are usually some reserved but unused ports in some scheduled conferences. Because these ports do not exist physically, there is no limit to the number that can be configured.
Scheduling
The audio conference can be scheduled through any of the following interfaces:
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Audio Conference Cascading
Conference cascading is also known as a multiserver meeting. This feature provides a virtual link between different MeetingPlace systems so that users on each server can communicate with each other as if they were in the same meeting. When users schedule multiserver meetings, they use the web schedule on the primary server to select the secondary servers required for the meeting. At the start of the meeting, the primary server places a call to the secondary servers. The secondary servers then add the primary server to the meeting, just as they would with any participant who dialed into their system.
You can cascade up to three Audio Servers for a single meeting.
Cisco recommends configuring Network Time Protocol (NTP) servers on all MeetingPlace Audio Servers to synchronize time across all the servers.
For more information on multiserver meetings, refer to the Administration Guide for Cisco Unified MeetingPlace Audio Server, available at
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_maintenance_guides_list.html
Dialing into an Audio-Only Conference Using Video Endpoints
Video endpoints can join audio-only conferences by dialing into the MeetingPlace Audio Server directly. MeetingPlace supports both out-of-band and in-band DTMF digits from the endpoints. When using the outdial feature from the Web server, a user must enter the video endpoint extension in the audio endpoint phone number box.
Tip
Web Conferencing
Web conferencing involves the following MeetingPlace components:
MeetingPlace Web Server
The MeetingPlace Web application provides the following primary functions:
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This feature provides scheduling, meeting room, meeting controls, reference center, and account management.
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This feature provides collaboration, whiteboard, application sharing, file uploads, and more.
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If MeetingPlace Web is installed with the audio service option, it first converts MeetingPlace voice (.mpv) files to .wav format, then you can choose to convert them to one of the other supported audio formats, such as Windows Media (.wma), RealAudio (.ra or .rm), and MP3. These files can be downloaded and made available to other applications such as a custom website or a CD.
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The .wav file is used by MeetingPlace Web to create a synchronized audio/web recording if the optional voice and web recording is enabled on the system.
MeetingPlace Web requires a Media Convergence Server (MCS) with the IIS service running, and it uses a Microsoft SQL database that is automatically synchronized with the MeetingPlace 8100 Audio Server master database.
The Cisco MCS 7835 server supports up to 50 simultaneous user sessions and the MCS 7845 supports up to 200 simultaneous user sessions, but there is a limit of 150 user sessions per meeting. In addition, there is a limit of 100 simultaneous web conferences for application sharing or 150 for presentation mode on a single Web server. (Presentation mode involves an uploaded presentation that is converted to JPEG format and downloaded individually to the local browser cache as each participant joins the web conference.)
Deployments with more than 100 Web Conferencing user licenses should not use Microsoft Desktop Engine (MSDE) 2000 because it is limited to eight concurrent connections for scheduling and web conferences. SQL 2000 is required and must be provided by the customer. Large installations (more than 500 user licenses) should run SQL 2000 on a dedicated server.
To increase performance in terms of capacity and load balancing, you can group MeetingPlace Web servers into clusters. The MeetingPlace Web cluster can support a maximum of six Web servers in either internal or external configurations.
Note
Clients can connect to MeetingPlace Web via either HTTP or HTTP over Secure Socket Layer (HTTPS). The following destination ports are used on the MeetingPlace Web server:
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SQL Database
The Microsoft SQL database on the Web server includes profile information as well as meeting information. The type of meeting information depends on whether the Web server is internal (hosting all internal and external meetings) or external (hosting only external meetings). The primary database is on the Audio Server. When a meeting is scheduled, the meeting information is also copied to the appropriate Web server to facilitate certain functions without having to go through the Audio Server for all requests.
Beginning with MeetingPlace 5.3, the database contains the following data:
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