Ip voice transmission, Speech encoding methods, 4 ip voice transmission – 2N VoiceBlue Next v3.0 User Manual

Page 29: Ip voice transmission 2.4

background image

IP Voice Transmission



2.4 IP Voice Transmission

Speech Encoding Methods

Voice transmission is strictly separated from signalling in VoIP networks. Modern VoIP
networks mostly use the RTP (Realtime Transport Protocol) for voice transmission. The
purpose of the RTP is only to transmit data (voice) from a source to a destination at
real time.
Codecs are used to save the channel data capacity. Codecs process the voice signal
using variable algorithms to minimise the volume of user data. The degree of
compression used by the codec affects the quality of voice transmission. Thus, the
better voice transmission is required, the wider data range (the higher transmission
rate) is needed. The MOS (Mean Opinion Score) scale is used for rating voice
transmission quality, where 1 means the worst and 5 the best quality. For a survey of
the codecs supported by 2N


VoiceBlue Next refer to the table below.

Codecs supported



Transmission rate [kbps] MOS














For 2N


VoiceBlue Next, quadruple the above mentioned rates (two fully duplex calls)

and add the TCP and IP header transmission rate to the result to get the resultant
transmission rate.

It is important to keep both a stable appropriate transmission rate during connection
and a small and identical transmission time per data packet in order to maintain a
high-quality voice transmission.

G.711 – this codec is used in digital telephone networks. The PCM (Pulse Code
Modulation) is used for voice signal encoding. The sampled signal is encoded
in 12 bits and then compressed using a non-linear scheme into the resultant 8
bits. Europe uses the A-law compression system while North America and
Japan obey the

-law. The resultant data flow is 64 kbps.

G.729 – this codec uses the CS-ACELP (Conjugate-Structure Algebraic-Code-
Excited Linear-Prediction) algorithm with the resultant transmission rate of 8
kbps. The speech signal is split into blocks of 10 ms each. The parameters of
these blocks are then inserted in frames of the size of 10 bytes. 2-byte
frames are generated for noise transmission.

During call set-up, a codec is selected automatically for voice transmission. 2N


VoiceBlue Next supports the codecs included in the table above.

The type of codec to

be used depends on your VoIP network (individual devices) and your 2N



Next configuration. 2N


VoiceBlue Next is designed primarily for VoIP corporate

networks and tries to meet the opponent‟s codec requirements. If a codec is requested
that is incompatible with 2N


VoiceBlue Next, the call will be rejected.


G.729 is an optional part of the system.