ASCOM TD 92326GB Installation And Operation Manual page 145

Installation and Operation Manual
VoIP Gateway
Voice encoding schemes
Encoding
G.711A
G.711U
G.726-32
G.723-53
G.729A
a.The specified bandwidth is merely the nominal bandwidth of the encoding algorithm. Additional
control information is transmitted in the network together with the compressed data, with the ef-
fect that, depending on the configuration, the total bandwidth required may turn out to be con-
siderably higher.
b.This is the minimum delay caused by data encoding and packeting. Further delays occur in con-
nection with the transmission of data in networks.
c.You can use both µ-law and A-law encoding, regardless of the encoding used on your ISDN con-
nection. In both cases, the encoding is correctly adapted to the ISDN connection.
If the remote VoIP device does not support the selected encoding, encoding supported by
both parties will be negotiated. Select the exclusive check box if you want to force the use
of the selected encoding. This can of course result in call failure if the VoIP Gateway and
the remote VoIP device do not support a common Coder.
Tip: The best trade-off between voice quality and required bandwidth is offered by
G.729. Select this scheme for remote telephony gateways accessed via the
Internet, the intranet or heavily loaded local area networks.
Use G.711 in powerful local networks, to ensure best voice quality.
You need G.723.1 for connections to telephony gateways which do not support
the G.729 standard.
G.726 encoding should only be used in cases where fax data is to be transmitted
on a line without T.38.
Packet size
You can set the size of the packets used for exchanging encoded voice data between
telephony gateways in Frame. The value defines the period of time for collecting voice
data prior to transmitting it as a voice data packet. Voice transmission is delayed
correspondingly. A value of 30 ms is perceived by the human ear as virtually without delay,
a value of 100 ms similarly, does not irritate most users.
Larger packets cause greater delays in voice data transmission, but cause less stress to the
network since the overhead involved in transporting packets in the network is lower.
Note that the overhead is increased considerably if the packet size is reduced, since the
overhead data required for transmission with the IP-protocol (on a LAN) and also in the
2009-03-10 / Ver. E
Bandwidth Minimum
a b
per call delay
64 kbit/s 20 ms
64 kbit/s 20 ms
32 kbit/s 20 ms
5.3 kbit/s 30 ms
8 kbit/s 20 ms
Properties
No compression, best voice quality
(comparable to digital telephone
systems). Sound digitising using
European encoding
As above; sound digitising using US
c
encoding
Intended only in exceptional cases for fax
and modem data.
Good voice quality (comparable to
analogue telephone systems)
Best voice quality of all compression
encoding schemes, lowest minimum
delay.
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