In previous articles you learned that a traditional voice call over the PSTN uses 64 kbps of bandwidth per call over a dedicated circuit-switched connection. While packet-switching solutions like VoIP do not require a dedicated circuit to operate, they still require sufficient bandwidth in order to function in a manner acceptable to users. Recall from earlier that a significant portion of any phone conversation involves periods of silence. In the world of packet-switched voice, those periods don’t explicitly require any packets to be sent (techniques like Voice Activity Detection or VAD can be configured using software like Cisco CallManager), which can provide bandwidth savings (typically in the range of 30-40%). However, one other significant method is also used to control the amount of bandwidth required in a packet-switched voice implementation – compression.
In much the same way that various compression schemes looked at earlier in this chapter can be used to reduce the size of data packets, compression schemes are also used on a voice network. Through the implementation of different codecs (coders/decoders), different levels of compression for voice traffic can be achieved, each with varying levels of perceived quality to the listener. This is a key consideration when implementing a voice networking solution – while the highest compression rate may produce the greatest bandwidth savings, it may also produce lower overall quality (this is not necessarily the case, as you’ll see shortly). Furthermore, the use of codecs that compress traffic to a higher degree can also add additional delay to the voice network.