NexLog Recorder User Manual v2.2.0
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transmitted using a single, static pathway over the telephone network. It uses
the Public Switched Telephone Network (PSTN), which is a circuit-switched
network, meaning the connection between the endpoints (telephones) is made
through switches that connect the lines together.
On the other hand, VoIP transmits the call using a packet-switched network.
With VoIP, the audio signal of the telephone call is digitized and encapsulated
into data packets that are sent over the network to the other party. The packets
may take one or more paths over the network to reach the called party. At the
other end of the line, the packets are reassembled and converted back into
analog voice signals. This network can be used at the same time by other
communications, which may include other VoIP telephone calls as well as a
variety of packetized information such as data and video.
Because the VoIP network can carry many conversations at the same time and
because it can also transmit other types of information, VoIP is a more efficient
and flexible method for transporting voice. It can also produce a richer
experience for the user if it is combined with other features, such as video. In
addition, it can be cost-effective to implement because you may be able to add
VoIP telephony services to an existing network infrastructure.
VoIP systems can interconnect and co-exist with existing PBX systems as well
the traditional circuit-switched network. Of course, power sources are a
consideration when implementing any VoIP system, because VoIP phones do not
derive power from a PBX or from the telephone company Central Office. So, to
protect against loss of telephone service due to power outages, it is necessary to
install uninterruptible or back-up power supplies for both the LAN equipment
and VoIP telephones.
Technical Considerations
The handling of audio data in VoIP differs significantly from how it is done on a
conventional, circuit-switched network. On the latter, once a connection is
established, it is defined between two fixed points, and both the upstream and
downstream data are handled by the same pair of wires. The digital architecture
of VoIP separates upstream and downstream data, and the transmission path
across the network can vary. Audio is carried through RTP (Real Time Protocol)
packets, which can be routed along different paths. As a result, data packets of
audio data can become unsynchronized and be delivered out of their original
sequence.
To address this, VoIP uses a buffering system that synchronizes delayed
packets. The inherent delay caused by packet buffering should never exceed 500
ms.
Networks are by no means limited to carrying only voice data. As such, a packet
filtering mechanism is used to detect and isolate RTP audio data packets from
other data types carried across the network.
