• devices that create and manage a SIP session, such
as a radio-to-VoIP gateway or a SIP VoIP phone,
• servers, such as a Registrar Server that stores device
registration information in a database, and Proxy
Server that re-routes requests or messages. (A SIP
server may be both a registrar and a proxy server at
the same time.)
Real Time Protocol (RTP)
RTP provides data transport for audio and video over
IP in telephony, video teleconferences and television.
Designed for end-to-end, real-time transfer of
stream data, it prioritizes “real time” over reliability,
often compromising audio/video quality for
RTP can send voice and radio information, but requires
more system configuration and is less flexible to operate.
However it can be used for unicast or multicast links,
and can manage issues that occur when sending data
over packet-switching networks. For example RTP can
detect lost data packets, and can rectify packets arriving
in the wrong order, or with variable packet delivery
delays (packet jitter).
Common VoIP Issues
Two types of delay occur within packet switched
• constant delays or “latency”
• variable delays or “jitter”
To compensate for jitter, the end device keeps a buffer
of audio data so that it can continue to play audio even
if the next packet is late. This in itself can be another
source of delay.
Radio PTT means that these systems tolerate delay
better than telephone systems, provided PTT signalling
is synchronized with audio signalling so that no leading
or trailing syllables are cut off.
Without delay, echo is not usually an issue for system
users. Once a delay is added, the echo may be audible.
Echo can be eliminated using Digital Signal Processors
(DSP) to process the audio, where echo cancellation
algorithms filter the echo signal from the received audio.
Sometimes data packets don’t make it to the receiver.
Fortunately, digital voice is generally intelligible
with quite high levels of packet loss as VoIP systems
incorporate Packet Loss Concealment (PLC) algorithms
to compensate. Well-designed RoIP systems can provide
acceptable, intelligible audio with packet loss of 10%.
On wired networks (LANs or WANs), packet loss
generally only occurs as a result of overload or
congestion. However, WIFI networks or Microwave links
are subject to loss of individual packets.
System designers and operators need to be aware that
their communications may depend on other devices that
may be less tolerant of packet loss.
A word about CODECs
Codecs (Coder/Decoders) transport voice through
packet switched data networks. Software codecs take
digitized audio and encode it to send over the network,
compressing the audio, so less data is required to
transmit it. Codecs for VoIP compress by discarding
some audio information, potentially reducing
For real time communications like VoIP, codecs need to
process audio in real time, so it only has 20 milliseconds