Inter-Tel AXXESS Manual page 166

Specifications
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INTER-TEL AXXESS
C. PROCESSING LATENCY
7.15 The amount of latency introduced by a IPRC depends on the specific vocoder being
used. When calculating latency, you must consider two vocoder-specific variables and one pro-
grammable parameter, as described below.
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•
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7.16 The basic formula for calculating end-to-end latency is as follows:
Latency = ([AF + 2] * FS) + MPBT + estimated one-way network delay
Where the estimated one-way network delay is based on dividing the round-trip ping time by
two.
NOTE:
tion in network delay.
7.17 The following is an example of how the total end-to-end latency value is calculated.
This example uses the default settings (G.729, three audio frames per IP packet, and 80 ms
buffer time) and assumes a 20 ms one-way network delay.
Latency = ([3 + 2] * 10) + 80 + 20 = 150 ms
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MANUAL VERSION 11.0 – May 2008
Frame Size (FS): Is a fixed number and is inherent to the specific vocoder being used.
The following table distinguishes the Frame Size for each vocoder in milliseconds.
Audio Frames per IP Packet (AF): Is a programmable parameter (Audio Frames Per
IP Packet; see page 6-185) for each Node IP Connection Group that indicates how
many vocoder frames are packed into a single packet. For minimal latency (but more IP
or Frame overhead), configure the system with one audio frame per IP packet. For
lower overhead, use larger values for this field. By default, a value of three is used for
the Audio Frames Per IP Packet field because it provides a good balance between low
latency and low overhead. You can configure this value while in the Network Qualifier.
Minimum Playback Buffer Time (MPBT): Defines the average time, in milliseconds,
that packets wait in the receive buffer before the system plays the audio. The higher this
number, the more latency in the signal; however, it is less likely that network problems
like jitter causes lost or late audio packets. The lower the minimum playback time, the
less latency there is in the signal; however, there is a greater chance of jitter affecting
the call. The default is 80 ms, and the range is 1 to 320 ms. You can configure this value
while in the Network Qualifier.
The actual end-to-end latency on a live network may be higher due to normal fluctua-
Table 2-22.
Frame Size
Vocoder Frame Size
G.729 10 ms
G.711 10 ms
Processing Latency