Voice Over Internet Protocol 101

Prepared by CTI Communications, LLC
Revised April 14, 2006

What You Need to Know Before Implementing VoIP

Voice over Internet Protocol (VoIP) is a technology that has made rapid advancements over the past few years, allowing phone conversations to be converted into IP packets (a digital series of 1’s and 0’s) and transported over any IP network, such as a Local Area Network (LAN), Wide Area Network (WAN), ISDN/DSL/Cable Internet, Frame Relay, Point-to-Point T1, ATM, and several others. Properly configured, VoIP provides call quality that is equal or better than traditional phone lines while offering substantial cost savings and tangible benefits (i.e. extension-to-extension dialing, remote administration, etc.). Our typical customer enjoys a full ROI of their VoIP implementation in less than one or two years.

For most customers, the biggest advantage of VoIP is the possible toll savings. Generally, the IP network incurs a flat monthly fee no matter how much traffic passes across it and regardless of geographic distances (i.e. it costs the same to send an IP packet to/from Denver as it would be to/from New York). Conversely, long distance charges are billed in per minute increments and can be very expensive depending on how many calls are placed and the duration of the long distance call. The “Hop Off” feature allows calls that normally incur long distance charges to be placed as local calls to/from other calling areas where you have an AltiGen system that is connected via VoIP.

Another benefit of VoIP is the ability to support remote users. Employees working from home or a branch office can be seamlessly integrated with the AltiGen system – they are essentially just another extension with the exact same features and functionality as an extension located within the office. This means that a VoIP extension can be a member of an ACD Workgroup, have a DID Number, Dial Other Extensions Directly, and Configure Call Handling Options – a VoIP Extension can do anything that a physical analog extension can.

There are two different methods to implement VoIP using the AltiGen, “Server-to-Server” and “Server(s) with Remote VoIP Extensions.” The first option is used to connect two or more offices together via VoIP, where each office requires the ability to have its own local phone lines. The second option is used to place individual VoIP phones at remote locations, such as an employee’s home or at a branch office where local phone lines do not need to be connected. Please see the brief diagrams below for clarification:

Here are the 5 basic requirements your IP network must be able to meet in order to be a candidate for VoIP:

• An IP address for each AltiGen Server and VoIP Extension. Although a static IP address is recommended for the AltiGen server(s), either static IP addresses or DHCP can be used for VoIP extensions. With AltiWare OE 4.5a, Network Address Translation (NAT) is supported, provided that the remote VoIP extensions have an H.323 NAT capable router installed (note that NAT locations are limited to a single VoIP extension each). For remote VoIP extensions, we currently recommend the Linksys DSL/Cable Router Model #BEFSR81 that supports both H.323 NAT and QoS. Please consult with one of our technical sales engineers for the network details regarding VoIP prior to your implementation.
• The AltiGen supports the industry standard H.323 protocol, and both the G.711 and G.723 forms of compression. Each concurrent call requires 75k of dedicated bandwidth for G.711 (toll quality) and 17k of dedicated bandwidth for G.723 (near toll quality). Bandwidth should be allocated to provide sufficient capacity for the maximum number of possible calls at any given time.
• The connection between VoIP connections (either Server to Phone or Server to Server) should have latency (ping times) of less than 100 ms (120 ms maximum) and minimal packet loss (never more than 5%). Use of the public Internet is generally not recommended for this reason, but if it is to be used, we suggest the use of the same ISP for both the host and remote locations to reduce the number of network hops.
• Quality of Service (QoS) to Prioritize VoIP Traffic Over Data. This can usually be accomplished at the router level by implementing port prioritization, but more complex implementations may require the use of a packet shaper. QoS is an integral part of VoIP and without it, your call quality may suffer since IP bandwidth is dynamic. QoS may be required at the host location and all remote locations, depending upon the configuration and bandwidth usage.
• Firewall Access – If VoIP traffic travels through a firewall or NAT’ed router, then the following ports on the firewall or router must be opened and forwarded to the AltiGen server and to any remote IP phones: TCP 1720, UDP 1718-1719, UDP 2222-2242 and both TCP and UDP from 49152 through 49771. Note: the end range is determined by the following formula: (49152 + (62 x # of VoIP Boards) –1). 49771 assumes that the current maximum of 10 VoIP boards permitted under OE 5.0A are being used. TCP port 10032 must also be opened to support remote AltiGen IP-600 & IP-710 phones. TCP 30040-30042 and UDP 30000-30001 must also be opened to support the VoIP protocol. Also TCP 49151 must be opened to support the message waiting indicator light on the IP-600 and IP-710 phones. For IP-600 & IP-710 firmware upgrading UDP 69 (TFTP) and UDP Ports 10040 - 10049 must be opened as well to allow the firmware image to pass through the firewall or NAT’ed router.

Please consult with one of our technical sales engineers for the network details regarding VoIP prior to your implementation. Unless specifically stated otherwise, the customer is fully responsible for providing, maintaining and troubleshooting the necessary network infrastructure outlined above to support VoIP, including but not limited to, LAN/WAN data circuits, routers, switches/hubs, firewalls, packet shapers, network interface cards, and all similar devices.