Internet protocol telephony

Last Updated : Dec 12, 2023 |

Internet Protocol (IP) refers to the type of rules that the network uses to send and receive signals. IP telephony converts voice communications into data packets. Conveniently, it runs on Ethernet LAN (local area network) technology. IP telephony unites a company's many locations, including mobile workers, into a single converged communications network.

IP Office is a converged telephony system; it combines aspects of traditional PABX telephony systems and IP data and telephony systems. IP Office supports PSTN, SIP, POTs, digital time division multiplexed telephones and digital IP telephones all on the same system. IP Office allows all the technologies to coexist.

IP Office connects to the PSTN and to IP trunks providing a hybrid PABX function - where both legacy and future technologies can be used together to minimize operating costs and optimize business communications through both voice and data. The converged functionality works on multiple levels:

Individual phone users can control the operation of their phone through applications running on their PC.
Data traffic can be routed from the LAN interface to a telephony trunk interface.
Voice traffic can be routed across internal and external data links. This option is referred to as voice over IP (VoIP).

Voice over IP (VoIP) and network assessments

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

Signaling protocols

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.

Quality of Service considerations

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.

Voice compression channels

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.

Table 1: Voice compression channels
Call type	Voice compression channel usage
IP device to non-IP device	Requires a voice compression channel for the duration of the call. If no channel is available, busy indication is returned to the caller.
IP device to IP device	Call progress tones (for example dial tone, secondary dial tone, etc) do not require voice compression channels with the following exceptions: Short code confirmation, ARS camp on and account code entry tones require a voice compression channel. Devices using G723 require a voice compression channel for all tones except call waiting. When a call is connected: If the IP devices use the same audio codec no voice compression channel is used. If the devices use differing audio codecs, a voice compression channel is required for each.
Non-IP device to non-IP device	No voice compression channels are required.
Music on hold played to IP device	Provided from the TDM bus and therefore requires a voice compression channel when played to an IP device.
Conference resources and IP devices	Managed by the conference chip which is on the TDM bus. Therefore, a voice compression channel is required for each IP device involved in a conference. This includes services that use conference resources such as call listen, intrusion, call recording and silent monitoring
Page call to IP device	Uses G729a for page calls, therefore only requiring one channel but also only supporting pages to G729a capable devices.
Voicemail services and IP devices	Treated as data calls from the TDM bus. Therefore calls from an IP device to voicemail require a voice compression channel.
Fax calls	These are voice calls but with a slightly wider frequency range than spoken voice calls. IP Office only supports fax across IP between IP Office systems with the Fax transport option selected. It does not currently support T38.
T38 fax calls	IP Office supports T38 fax on SIP trunks and SIP extensions. Each T38 fax call uses a VCM channel. Within a Small Community Network, a T38 fax call can be converted to a call across an H323 SCN lines using the IP Office Fax Transport Support protocol. This conversion uses 2 VCM channels. In order use T38 Fax connection, the Equipment Classification of an analog extension connected to a fax machine can be set Fax Machine. Additionally, a new short code feature Dial Fax is available.

Note:

T3 IP devices must be configured to 20 ms packet size for the above conditions to apply. If left configured for 10 ms packet size, a voice compression channel is needed for all tones and for non-direct media calls.

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