Monitoring and Managing Voice over Internet Protocol (VoIP)

VoIP phones use codecs to translate analog sound streams into digital packets for transmission. On the receiving end, the codec translates the packets back to analog. For two people to converse normally, all of this must happen in as close to real time as possible.

For call setup, most enterprise VoIP solutions include one or more call managers, which are servers that set up calls between VoIP phones, and can also provide gateway connections to the Public Switched Telephone Network (PSTN) for calls outside the VoIP network. Typically, the call initiator contacts the call manager, which then rings the phone being called. Once the receiving party answers, the call manager provides a mechanism for the phones to negotiate codecs and connection parameters.

The connection itself is typically in the form of two full-duplex streams: a Real-time Transport Protocol (RTP) stream that carries the encoded audio, and an RTCP (Real-time Transport Control Protocol) stream to provide communications control. Once the call is set up, the call manager is no longer involved until the teardown phase, when the IP phones inform the call manager the call has been completed so the centralized call queue (a list of what phones are active) can be updated.