4 ip voice transmission, Speech encoding methods – 2N BRI Lite/Enterprise v1.8 User Manual

Page 24

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

gateway refer to the table below.

BRI

Codecs supported

Standard

Algorithm

Transmission rate
[kbps]

MOS

G.711a

PCM

4.1

G.711u

PCM

4.1

G.729 G.729 is an optional part of the
system.

CS–ACELP

3.92

For

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

2N BRI

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 µ-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 BRI

ateway supports the codecs included in the table above. The type of codec to be used
depends on your VoIP network (individual devices) and your

gateway

2N BRI

configuration.

gateway is designed primarily for VoIP corporate networks and

2N BRI

tries to meet the opponent's codec requirements. If a codec is requested that is
incompatible with

, the call will be rejected.

2N BRI Enterprise

The SIP and ITU-T H.323 recommended protocols are mostly used for connection
establishing, maintaining and cancelling.

gateway uses the

(Session

2N BRI

SIP

Initiation Protocol) signalling.