6 telephony services – 2N OfficeRoute - User manual, 1493 v1.9.0 User Manual
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3.6 Telephony Services
Before the description of all items of the Telephony services menu, a short introduction
about the VoIP Network Arrangement and Routing behaviours is given there.
VoIP Network Arrangement
VoIP telephony communication has two components – signalling and voice. Signalling is
primarily used for establishing and terminating calls, telephone login to a PBX,
negotiating parameters and control of the speech/voice channel. The voice channel is
only used for transmission of encoded digitised voice information.
Typically, VoIP telephones are operated together with a PBX, which coordinates their
traffic. For a schematic arrangement see fig.39. The VoIP telephony PBX is a software
application that looks like a traditional PBX. Its functions include mainly numbering
plan consistence maintenance, routing, user or telephone rights, call billing, call
forwarding, DISA, etc. But it can integrate more functions, e.g. voicemail. SIP
telephone PBXs are usually called SIP proxies while PBXs for H.323 are called
gatekeepers.
Figure 3.30: VoIP PBX Arrangement
To work properly, a VoIP PBX must process all signalling traffic. Unlike this, voice
channel data are transmitted directly between the terminal points. This is, among
others, a difference from the model described below. It is because a PBX is not
necessarily required in the IP telephony. Signalling protocols are designed in such a
manner that it is possible to call from one terminal to another directly, without any
mediator. To do this, you have to know the full ID of the terminal to be called, i.e. also
the IP address and destination port, of course. This is the main disadvantage compared
with the preceding model for multiple-telephone locations.