VoiP codecs have been created to serve the primary function
of optimizing voice over IP networks. Yet, there are other codecs specializing
in music-video transmission.
The quality as well as the features attributed to any VoIP applications or
devices affect to a great extent the general quality of Internet calls. Like
already mentioned at the beginning of this article, codecs take over our
voice compression-decompression-which, in the meanwhile, has been digitalized-,
that it occupies the least possible volume, while it is being transmitted
in the Internet. At the same time, they maintain voice quality at satisfactory
To make a VoIP call feasible, both parties have to not only adopt the same
communication protocol, but the same codec as well.
The main codec features are as follows:
- Sampling rate: 8 to 16 Khz are more than enough to achieve quality while
- Codec bit rate: In this case rating fluctuates between 5 and 64 kbps.
When referring to data flow, it is widely accepted that what is meant focuses
on the rate of
compression performed by the codec. A high rate means little or zero compression,
therefore higher sound quality. On the other side, low compression is responsible
for a big volume of packets and as a result, an increase in delays and
- Nominal bandwidth: It refers to the bandwidth needed for our connection
with the Internet or in a LAN, so that the codec is able to achieve maximum
performance. Apart from each codec’s data flow, the “load’ of
various protocols being in control of the flow of packets like the UDP, RTP,
and others, also
has to be counted in, in the total bandwidth required.
- Time required
to the creation of voice packets: The more time required to create voice
packets, the less network load is achieved. On the contrary,
mean that the codec will have to compress a rather large volume of sound
data, resulting thus in delaying data transmission as well as their reception.
acceptable compression times range between 10 and 40 ms.
MOST WIDESPREAD CODECS
G.711. It is the codec being used by digitalized fixed telephony networks.
It relies on the Pulse Code Modulation [PCM] and provides relatively
low compression. This means high quality when it comes to voice transmission;
yet, wide bandwidth
is required while connection is being attempted.
G.723.1. The high compression achieved by this particular codec relates
to a minimum bandwidth to achieve VoIP communication. However, the quality
accomplished reaches mediocrity, while delays in telephone conversation
increases. The fact that it has been designed to accommodate tele-conferences
communications carried out by plain telephone lines is more than obvious.
G.729. It is normally found in VoIP SIP applications. It provides double
compression than the one offered by the G.711, meaning that a rather
high voice quality
can be achieved, though the times required, to prepare coded voice packets
GSM. None other than the quite well known and qualitatively tested codec
applied to the GSM mobile telephony networks.
ILBC. Its full name is the portrait of its capacities, [Internet Low
Bitrate Codec]. It is provided at no cost, though some limitations are
by the Global IP Sound. It is designed to facilitate use in low speed
networks and incorporates special voice improvement techniques for lost
Speex. It is an open software codec with very good technical features
when compared to several “restrictive” as well as expensive
Skype. Last but not least, comes the codec referring to the most popular
VoIP application. It has been developed by the Global IP Sound and
into exclusive utilization by the Skype.