Voice over IP (VoIP) means voice transmitted over a packet data network. VoIP is often referred to as IP telephony because it uses the internet protocols to make possible enhanced voice communications wherever IP connections exist.

The VoIP mode of operation can include external SIP trunks, IP trunks between customer systems and/or H.323 or SIP IP telephones for users. In either case the following factors must be considered:

• The IP Office control unit must be fitted with voice compression channels. These are used whenever an IP device (trunk or extension) needs to communicate with a non-IP device (trunk or extension) or to a device that uses a different codec.
• A network assessment is a mandatory requirement for all systems using VoIP. For support issues with VoIP, Avaya may request access to the network assessment results and may refuse support if those are not available or satisfactory.

A network assessment would include a determination of the following:

• A network audit to review existing equipment and evaluate its capabilities, including its ability to meet both current and planned voice and data needs.
• A determination of network objectives, including the dominant traffic type, choice of technologies, and setting voice quality objectives.
• The assessment should leave you confident that the implemented network will have the capacity for the foreseen data and voice traffic, and can support H.323, DHCP, TFTP and jitter buffers in H.323 applications.

An outline of the expected network assessment targets is:

Test	Minimum assessment target
Latency	Less than 150 ms
Packet loss	Less than 3%
Duration	Monitor statistics once every minute for a full week

In order to make use of VoIP, IP Office uses signaling protocols called H.323 and Session Initiation Protocol (SIP), to establish end-to-end connections for the voice path through an IP network. This connection ensures that each end is able to transmit and receive voice and provides the network addressing for end-to-end packet transmission. IP Office also connects the different technologies by translating the signals they use. For example an analog telephone can connect to a VoIP destination. This connection requires both the signaling and voice transmission to be translated. IP Office makes this translation using gateways and gatekeepers.

With IP telephony you connect an IP telephone to an IP PBX through a LAN. There are two basic types of IP telephones:

• A physical telephone, which looks very similar to a standard telephone, referred to as a hardphone
• A software application, referred to as a softphone, which runs on the user’s PC, allowing them to use a headset and microphone to make and receive calls anywhere they an and IP connection.

When making use of IP telephony, there are a number of data centric considerations such as which data types have priority on the IP network when there is contention. This is set with IP/TCP quality of service and should not be ignored. In situations where LAN bandwidth is limited, a quality of service capable LAN switch should be used to ensure voice packets are transmitted with the required priority on the network. If not, the conversation carried over IP appears as broken up due to delays or has unacceptable delays introduced in the conversation causing latency and jitter. With IP hardphones there is the need for Power over Ethernet (PoE), or local phone power supplies to be provided to the telephones as the IP telephones are not powered by IP Office.

Calls to and from IP devices can require conversion to the audio codec format being used by the IP device. IP Office systems use voice compression channels to make the conversion. These channels support the common IP audio codecs G.711, G.723 and G.729a.

The System Status Application can be used to display voice compression channel usage. Within the Resources section it displays the number of channel in use. It also displays how often there have been insufficient channels available and the last time such an event occurred.