Introduction to Basic IP Telephony Call using CUCM

 Today I am going to talk about the basic IP telephony call using CUCM. CUCM stands for Cisco Unified Communications Manager. At the start of a call, an operator at IP phone A took up the handset, and a communication is sent to CUCM letting CUCM understand that the device has gone off. 

CUCM replies to this incentive by replying with a message that says the device to play the dial tone file that is kept in the flash memory of the phone. The user at phone A gets the dial tone and starts dialing the phone number of phone B. SCCP phones direct their digits to CUCM as they are pressed (digit by digit), while SIP phones direct their dialed digits in one message by default. 

Fig 1.1- CUCM based Basic IP Telephony Call

SIP phones have choices that permit them to act similarly to SCCP phones. CUCM achieves digit analysis beside the dialed digits. If a match is found, CUCM routes the call per its configuration. If CUCM does not find a match, a reorganize tone is sent to the calling party.

CUCM indications the calling party to initiate ring back, so the user at phone A will get the ring back tone. CUCM also indication the call to the destination phone, which shows the ring down tone. Added data is provided to the phones to specify the calling and called party name and number. (Phone A will display the destination device name and number, and phone B will display the calling party name and number.)

When the user at phone B receives the call, CUCM directs a communication to the devices allowing them know the IPv4 socket (IPv4 address and port number) data in which they should connect for the period of the call. The RTP media path unlocks directly between the two phones.

The Cisco IP Phones need no further announcement with CUCM until either phone raises a feature, such as call transfer, call conferencing, or call termination.