Voice over IP (VoIP) Fundamentals
101
A modern Intranet normally offers enough performance reserves and reliability to
make good-quality VoIP telephony possible. Specific components can also be
optimised; for example by using a modern switch which evaluates the TOS byte of
IP packets or by replacing unreliable connections.
9.1.2 Latency and Packet Length
For technical reasons, there is always a delay (“latency”) between the recording of
voice data via the microphone and playback via the receiver. Voice data is
recorded for a short period so that it can be sent in an IP packet. The IP packet also
has a signal-propagation delay before the receiver can begin playback. For these
reasons, the extra time required for voice-data encoding and decoding may be
neglected.
An IP packet consists of protocol data and user data. Sending shorter voice-data
packets causes the ratio between the user data and the protocol data to become
unfavourable and increases the bandwidth required. Sending longer voice-data
packets increases latency.
The length of the voice-data packets must therefore be adjusted to the require-
ments of the transmission medium. Shorter voice-data packets can be sent if a
direct ethernet connection exists. If an 64 kilobit/s ISDN line is to be used for trans-
mission, then longer voice-data packets should be used.
Longer voice data packages are generally used for SIP telephony over the Internet.
The following table provides an overview of the required bandwidth for a tele-
phone connection with various parameter settings. The values apply to half-
duplex ethernet; for full-duplex the values can be halved.
Required bandwidth (kilobit/s) with respect to Packet Length and Codec
Packet
Length (ms)
G.711 (not
compressed)
G.729A
c. 6.3 kilobit/s
G.723.1
6.3 kilobit/s
G.723.1
5.3 kilobit/s
20 180.8 68.8
30 51.2 48.0 45.9
40 42.4
50 37.12
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