The Fundamentals of Cisco Telephony Design
A primary business requirement of today's enterprises is to reduce capital and support costs while improving the effectiveness of the network infrastructure, security and management. Cisco has developed IP telephony solutions that address those requirements while allowing for migration from current PBX switches in a cost-effective manner.
Current PBX switches are expensive to purchase and support. IP telephony offers an alternative that replaces the company PBX with Call Manager software that will run on a Windows server. The current version of Call Manager has features such as unified messaging integration, call routing with fail over, easy migration from the current PBX infrastructure, and management that is integrated with CiscoWorks.
Since Call Manager was designed to replace your PBX, there are all of the telephony features that you would expect with any voice switch. It is important to distinguish between IP Telephony and Voice over IP, Voice over Frame Relay and Voice over ATM. The voice router modules interface with the existing PBX at each company office and don't use IP phones. IP telephony will replace the PBX switch at each office with Call Manager for switching and routing of all voice traffic. The data circuits are utilized with Call Manager as with Voice over IP. There is a capital expenditure with the purchase of IP Phones, which are designed to work with Call Manager software. There is also a cost-effective option called a Soft Phone, which runs as an application at the client desktop.
The employee requirements should be considered to determine which is most appropriate for the employee. As mentioned, there is diversity (fail over) built into IP Telephony from a circuit, call routing and IP Phone perspective. The current PSTN circuit can be utilized for fail over should the IP WAN circuit be unavailable. The PSTN circuit is configured to transport voice traffic during periods when the IP WAN circuit is congested or when it is unavailable. Call routing diversity is provided with Call Manager clustering, which designates a group of servers that continue processing calls when a server isn't available. The IP Phone can be plugged into a different campus switch port than the desktop or it can plug into the same port and assign separate VLANs to data and voice traffic. Plugging the phone into a different switch than what the desktop uses will prevent both phone and switch being unavailable should a campus switch fail. However, this approach can become very expensive.
Quality of service is important with any voice migration project since voice traffic is delay-sensitive. Bandwidth must be allocated to reduce or eliminate jitter. IP phones can prioritize voice traffic before it arrives at the campus switch. The campus switch examines each packet and will allocate bandwidth for packets that are designated as higher priority.
This design implements Call Manager at an office with hundreds of employees. This implementation has the desktop and IP phone sharing the same switch port, which is the least expensive option for connecting an IP phone to the network. There are three options for powering your IP phone. Some campus switches offer an in-line power module for powering the phones. If that switch type isn't available for your company, you can plug the phone into a 120-volt AC wall outlet using an adapter or purchase an external patch panel that sends power across pins 1 and 4 of the RJ-45 phone cable.
Call Manager is installed on a Windows server, which is connected to the switch. A Voice Mail server is connected to the Call Manager server for processing employee Voice Mail requests. Cisco offers a Voice Mail server called uOne. The switch connects to the voice-enabled 3660 router with a 100 Mbps Ethernet link. The router runs IOS voice gateway software, which provides E.164 Address Resolution and WAN call admission control. There are two available circuit options for failover design. The primary circuit utilized will be the data circuit, as it is with Voice over IP. The PSTN vendor usually charges the client by usage, which makes it important to provision your data circuit with additional bandwidth for the additional voice traffic. That will address the design requirements for performance while minimizing congestion and subsequent traffic across the PSTN.
As mentioned, voice packets can be prioritized at each IP phone for end-to-end quality of service at Layer 2 and 3 of the OSI model. Packet prioritization at Layer 2 is available with 3 bits of the 802.1p field of an 802.1q tag. 802.1q is a tag that is appended to each packet with the VLAN membership of that packet. When the packet is sent across a switch trunk line, each switch will read the 802.1q tag to determine VLAN membership for that packet.
The 802.1p priority field defined with the 802.1q specification is utilized for setting those 3 bits to a value between 0 - 6 that designates a higher priority for a packet from the default of 5. The switch will examine the 802.1p field and prioritize packets with a higher number over those with a lower number. Prioritization at Layer 3 is available with 3 bits of the Differentiated Services Control Packet (DSCP) field of the Type of Service (ToS) byte specified with each IP header. Those 3 bits are set to a value between 0 - 6 to indicate a preference for voice packets to be prioritized.
Each IP phone must request an IP address, subnet mask and default IP gateway address, which is available with DHCP services. There is an option as well to assign those items with static IP addressing, which requires configuration when the IP phone is implemented. The best option is DHCP since there is a lot less administrative work. The IP phone then registers their IP address and dial number with Call Manager before sending voice packets across the network. Cisco IP phones are designed to use G.729 compression for all voice traffic.
Once the traffic is compressed, it is sent to the connected campus switch. When it arrives at the switch, the 802.1p and DSCP fields are examined for prioritization. Those voice packets that are marked as priority are then processed by the campus switch before data packets. Call Manager examines the dial numbers and forwards the packet with the proper routing information to the router. The router will examine the DSCP field since it is a Layer 3 device and prioritize voice packets before data packets for transport across the WAN. The destination Call Manager server will examine the routing information of that voice packet and forward it to the campus switch where it arrives at the connected IP phone.
Call Manager has defined both distributed and centralized models, which define where the Call Manager servers are located across the enterprise network. The distributed model has Call Manager at each office and calls are routed between servers at each office. That model can support 100 offices with eight Call Manager servers per cluster. Each office can support 10,000 employees per cluster. The centralized model has one or more Call Manager clusters at a designated office, which requires that all calls be sent across the WAN to that office for routing between offices. A typical Call Manager cluster has three servers with the centralized model, which will process calls for 2,500 employees. It is possible to implement additional clusters at the centralized office for scalability as the company grows.