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Quality of Service (QoS) for Voice over IP (VoIP)
For VoIP the latency time, i.e. the time it takes for a voice packet to travel from sender
to recipient, is very important. Hence SX-GATE's traffic shaper optimizes delivery of
VoIP data packets with a special quality-of-service module.
In order to be recognized as VoIP packet, a data packet
needs to be tagged according to Diffserv-Code-Point Expedited-
Forwarding (DSCP EF).
The amount of bandwidth a single call requires depends on the codec that is used. The
codec defines the compression of a VoIP call. The more compression is applied the
less bandwidth is consumed, however also the quality decreases. The following table
shows the net bandwidth required by commonly used codecs. Some codecs are used
at different bandwidths. In this case the maximum bandwidth is given.
Codec
max. bandwidth (bit/s)
G.711
64000
G.722
64000
G.722.1
32000
G.723.1
6400
G.726
40000
G.728
16000
G.729
8000
GSM
13000
iLBC
15200
Max. number of concurrent calls
Enter the expected maxium number of simultaneous calls on this interface. It is used
to calculate the overall bandwidth that needs to be reserved for VoIP traffic. The value
"0" will disable this feature.
The total bandwidth of the link must not be exceeded.
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