Cisco IP Telephony Solutions Deployment
Guidelines and Caveats
Cisco IP Telephony Solutions
Cisco CallManager version 2.4
These guidelines cover the deployment of Cisco IP Telephony Solutions components that use Cisco CallManager version 2.4. To ensure TAC support and successful installations in this phase, it is very important that you review this document, and then communicate the guidelines to all customers as part of the design process. All proposals and installations must fall within these guidelines and customers must be made fully aware of the limitations described below.
If you have a situation where you are taking an order that does not fall exactly within these guidelines, then please contact your Account or Channel SE. In all situations a release document must be completed and approved before a Callmanager solution can be ordered, contact your Account or Channel SE.
Guidelines for Cisco IP Telephony Solutions
Proposals and Installations
1. Network topology. Single site with no current IP WAN connectivity for voice
2. Maximum number of IP Phones per site. 200
3. Failover limitations. If there were two Cisco CallManagers at the site, the failover would be approximately 90 seconds and could require manual replication of the programming on the backup Cisco CallManager. This failover applies to 12 SP+ and 30 VIP phones sets, AT, AS, DT-24+, and DE-30+ gateways. For IOS gateways to fail over, manual H.323 RAS configuration equired.
4. Cisco CallManager must run on approved hardware. Cisco CallManager must run as a standalone application on the Cisco Media Convergence Server 7830 or 7820 (MCS-7820 or MCS-7830) platform.
5. IP Phone power. Power to the phones is available only through a local AC power adapter.
6. IP Phone Network connections. IP Phones connect to a 10 Mbps switched or 10/100 Mbps switched Ethernet connection only.
7. DHCP addressing for “daisy-chained” PCs. A PC or workstation that is connected to the IP Phone obtains its IP address from the same address pool (subnet) as the phone if DHCP is used for addressing.
8. Possible voice degradation for “daisy-chained” IP Phones. When a PC or workstation is connected to an IP Phone so that they share a network connection, it is possible that the PC can cause some voice degradation in voice quality if both devices are being used at the same
time with high traffic loads or IP Multicast applications such as IPTV. In practice, however, it is difficult to achieve this degradation with normal PC applications.
9. No firewalls between IP telephony devices. There can be no firewalls/NAT between any of the IP Phone, Gateway, or CallManager components.
10. The network must be designed for average or higher use. The LAN/MAN network should be appropriately designed to handle expected data traffic bandwidth, deliver acceptable voice quality of service, and account for normally expected over-subscription. This guideline may dictate LAN/MAN hardware/IOS upgrade.
11. Homologation. Availability of the Cisco IP Telephony Solution internationally is subject to homologation timelines. See specific product descriptions on http://wwwin.cisco.com/ent/voice/sales/prices.html for current international availability.
12. Voice messaging. For voice messaging support, only approved and tested configurations, networked with SMDI are supported. Currently approved voice messaging
configurations:
• Cisco uOne (Unified Open Network Exchange) voice
messaging system is supplied with the Cisco CallManager
on the MCS-7830 server. Cisco uOne can support up to 100
mailboxes. Additional mailboxes require the purchase of an
additional server to run Cisco uOne. This server will be
released in the latter half of the first quarter of calendar
2000.
• Lucent/Octel Message Server 250 - See Lucent
Technologies Configuration Note 5405 (Rev A - 7/99)
13. Legacy PBX to DT-24+ and DT-30+ connections. Legacy PBX integration to digital VoIP gateway interfaces are supported via a digital ISDN PRI interface only. For example, DT-24+, DE-30+, or IOS gateways with PRI ISDN interface card/port adapter support. This guideline does not preclude integration to analog interfaces. Channel
associated signaling support is available for IOS gateways with H.323 signaling using MTP for supplementary services. Caveats with associated CAS implementations on these gateways still apply.
14. If a digital gateway is more than 30 meters from the PSTN demarcation point or the PBX, a CSU is required. The customer must supply a channel service unit (CSU) if the connection between a digital gateway and the PSTN demarcation point or PBX is more than 30 meters.
15. Supported applications. Only the following Cisco IP Telephony Solution applications are supported:
• Cisco ActivePhoneBook
• Cisco ConferenceBridge
• Cisco Messaging Interface (previously named SUMI)
• Cisco uOne voice messaging (only on MCS-7830)
• Cisco WebAttendant
• Media Termination Point (MTP)
The following applications are demo prototypes and should NOT be installed in a production environment. These applications are specifically excluded from formal support:
• Cisco Valet
• Cisco VoiceInbox
• Cisco VirtualPhone
16. H.323 restrictions. Client H.323 interoperability with Microsoft NetMeeting (2.X/3.X) is supported. Non-Cisco H.323 gateways have not been tested yet and are not supported.
17. 911/E911 limitations. Emergency/enhanced 911 locator service requirements may not be met in all states in the U.S. and Canada. The Cisco IP Telephony Solution properly routes 911 calls. However, delivery of ANI information to support Automatic Locator Information (ALI) for E911 Public Service Answering Points PSAPs is not supported.
18. Modem connection to Cisco Access AS-X is not supported. Modem operation when connected to Cisco Access AS-X gateways is not supported.
19. Lifeline service must be available separate from the Cisco IP Telephony Solution. Lifeline service should be provided by separate phone(s) connected directly to a central office line or to a separate PBX from which any user at the customer site would expect lifeline services to be available. When deployed as a stand-alone PBX replacement, a separate analog phone, connected to the central office is a requirement. This lifeline phone should be easily identified to all users within the customer environment.
A completed Order Authorization Form is required for orders that include
the MCS-7820 or MCS-7830.