Codec Bandwidth and Delay

Codec Bandwidth and Delay

When we are looking for the most efficient use of bandwidth, the aim should be to pack the
maximum number of speech samples into each single IP packet. However, this approach is
limited by the available network connection speed, which might require smaller data blocks.
This is shown clearly in the following example. On a 28.8 kbps modem-type trunk, an 86 byte
data packet needs 23.8 ms for transmission whereas on a 100 Mbps 100Base-T Ethernet
link only 6.8 μs are needed.

In both scenarios, additional time is needed for packetizing as the IP packet has to wait until
the last speech sample has been encapsulated before network transmission can start. For
example, including 3 additional speech frames in the same IP packet will produce a waiting
time of 30 ms with 10 ms more waiting time for every further speech sample.
Real-time IP applications, such as VoIP, are very delay sensitive. All delays - that is, speech
coding and compression, packetizing and transmission - have to be countered to keep the
overall delay time at a level where speech quality is acceptable.