Codecs, Bandwidth, Delay, Jitter And Jitter Buffer; Codec; Bandwidth - Avaya T3 Service Manual

Codecs, bandwidth, delay, jitter and jitter buffer

Codec

Both the IP telephone and IP Office must have an audio Codec. This codes the voice
signals from the microphone for transfer via the transmitting terminal, and decodes the
received audio signals sent to the loudspeaker/receiver of the terminal.
G.711 and G.729 are supported in the IP telephone.
Nominal data rate
Process
Codec generates its
output data with...
G.711A 64 kBit/s
G.711 ì 64 kBit/s
G.729 8 kBit/s

Bandwidth

•
Data flow depends on the Codec and the configured delay.
•
Voice data are not transferred via the packet-oriented TCP/IP protocol on a
continuous basis.
•
The Codec compiles the voice data into a packet during a specific period of
time, 10ms for the IP telephone.
•
IP packets consist of a variable percentage of user data and a fixed
percentage of management information.
•
For small packets the ratio between user and management data is very
unfavourable for the required bandwidth but there is only a short voice delay.
Header
20 bytes
The user data header is 40 bytes.
Packetisation
delay
Pass through the
Codec
1 ms
1 ms
25 ms
RTP-Datagram
IP UDP
Header
8 bytes
Section of data transfer
Short description
No compression, voice quality
comparable with ISDN, European
process for audio digitisation
As for G.711A, except a US
process for audio digitisation
Best voice quality of the
compressing process
RTP
RTP
Payload (user data
Header
8 bytes
12 bytes
General explanations
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