Aastra OpenCom 510 User Guide - Page 119

Latency and Packet Length, c. 6.3 kilobit/s

Page 119 highlights

Voice over IP (VoIP) Fundamentals 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. 11.1.2 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 requirements 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 transmission, 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 telephone connection with various parameter settings. The values apply to halfduplex ethernet; for full-duplex the values can be halved. Required bandwidth (kilobit/s) with respect to Packet Length and Codec Packet G.711 (not G.729A G.723.1 G.723.1 Length (ms) compressed) c. 6.3 kilobit/s 6.3 kilobit/s 5.3 kilobit/s 20 180.8 68.8 30 51.2 48.0 45.9 40 42.4 50 37.12 117

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Voice over IP (VoIP)
Fundamentals
117
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.
11.1.2
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