had the enjoyable opportunity over the last few weeks to jump in and do a basic Cisco voice install. It was about 110 phones, with the Cisco Unified Communications Manager Business Edition. This is a single box that includes Call Manager 6.1.3, and Unity Connection 2.1. It had to be done fast, and it had to be done right, so I stuck to mostly tried and true configurations.
Since this was a price sensitive design, we used the 2800 router to its maximum potential. The 2800 is an amazingly flexible piece of equipment; it can be configured to do large variety of things. Sometimes known as the Integrated Services Router, or ISR, it can be set up as a router, firewall, VPN, Voice Gateway, SIP session border controller, transoder, conference bridge, and survivable remote gateway, all at the same time, on the same box!
The call manager and unity connection install was straightforward, like punching out license plates. Set up media, device pools, partitions, calling search spaces, translation patterns, gateways, route filters, route patterns, etc. Scan then batch add the phones, set up voicemail and autoattendant call handlers, create expections, deal with the special people, and that’s it. Enough said about that.
The Cisco 2800 Integrated Services Router is used in this example to terminate a Multilink PPP bundle of four Internet T1’s, act as a firewall, provide media services to the Cisco call manager, act as an MGCP controlled analog gateway, and use Cisco Survivable Remote Site Telephony (SRST) to be the backup call processor to the main Cisco Call Manager.
SIP is ok with Network Address Translation as long as the firewall is capable of doing deep packet inspection and NAT’s all references to IP addresses. When I tried to NAT the inside interface of the firewall…it did not work so well. The remote SIP service provider was seeing private IP addresses in the SIP text, which does not make for good two way communications.
The Quality of Service setup on this example is fairly straightforward. Outbound is the standard Cisco MQS low-latency queuing setup, with a priority queue for voice and class based weighted fair queuing for the rest. Even though the service provider has said they prioritize inbound voice, I still set up inbound policing. Non-voice is limited to 4 Mbps, and anything greater than that will be dropped. Voice can use all of the bandwidth, so essentially there is 2 Mbps reserved for inbound voice. This is based on a calculation of 80 kbps for one G.711 call, so 2000 kbps gives us 25 concurrent voice calls, which should be plenty for 110 phones.