2. Factors Affecting Voice Quality - NEC UNIVERGE SV9100 Manual

Voice over ip
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telephony and videoconferencing, streaming audio, archival and messaging.
G.729A
This ITU-T recommendation describes the algorithm for coding of speech signals at 8kbps using CS-
ACELP (conjugate-structure algebraic code-excited linear prediction). This codec samples the
analogue signal at 8000Hz and uses a frame size of 10ms. This CODEC has a MOS score of 4.0.
G.729 is the most commonly used CODEC for SV9100 VoIP installations. This is due to the fact that
it offers high compression (and therefore low bandwidth) whilst maintaining good speech quality.
G.723
This ITU-T recommendation describes a very low bit-rate compression algorithm. The standard
describes two versions 5.3Kbps and 6.4Kbps. SV9100 uses the lower bit rate. This CODEC offers
low bandwidth speech transmission, but has a lower MOS score of 3.9. This CODEC is not
commonly used on the SV9100, but is particularly suited to low bandwidth WAN connections.
MOS. The mean opinion score (MOS) provides a numerical measure of the quality of human speech
at the destination end of the circuit. The scheme uses subjective tests (opinionated scores) that are
mathematically averaged to obtain a quantitative indicator of the system performance.
Note:
Not all CODEC's listed above are available for all applications.
1.5
Compatibility?
It should be noted that NEC does not guarantee that any third-party equipment will operate correctly
with NEC equipment.
1.2

2. Factors Affecting Voice Quality

2.1
Quality of Service (QoS)?
Quality of Service (QoS) is one of the most important factors for VoIP. The term refers to the
perceived quality of speech and the methods used to provide good quality speech transmission.
There are several factors that affect speech quality, and several mechanisms that can be used to
ensure QoS.
This section describes the problems that can occur and some possible solutions.
2.2
Latency (Delay)?
If at any point the usage on the network exceeds the available bandwidth, the users will experience
delay, also known as latency. In more traditional uses of an IP data network, the applications can
deal with this latency. If a person is waiting for a web page to
certain amount of wait time. This is not so for voice traffic. Voice is a real time application, which is
sensitive to latency. If the round trip voice latency becomes too long (250 ms, for example), the call
quality would usually be considered to be poor. Another important thing to remember is that packets
can get lost. IP is a best effort networking protocol. This means the network will try its best to get
your information there, but there is no guarantee.
Delay is the time required for a signal to traverse the network. In a telephony context, end-to-end
delay is the time required for a signal generated at the talker's mouth to reach the listener's ear.
Therefore end-to-end delay is the sum of all the delays at the different network devices and across
the network links through which voice traffic passes. Many factors contribute to end-to-end delay,
which are covered next.
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