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CUCM uses SIP or SCCP to communicate with Cisco IP Phones for call setup and
teardown and for supplementary service tasks.
After a call has been set up, media exchange occurs directly between the Cisco IP Phones
across the IP network, using the Real-Time Transport Protocol (RTP) to carry the audio.
CUCM is not involved in a call after the call has been set up. If the CUCM server were
unplugged during the duration of the call, users would not notice unless they attempted to
use a feature on the phone. CUCM is involved only in call setup, teardown, and features. If
the CUCM server that set up the call were down during a conversation, end users would
see a message indicating “CM Down, Features Disabled” on the LCD screen of the IP
phone.
Example: Basic IP Telephony Call
Figure 1-4 illustrates a user at phone A placing a call to phone B.
At the beginning of a call, a user at IP phone A picks up the handset, and a message is sent
to CUCM letting CUCM know that the device has gone off-hook. CUCM responds to this
stimulus by replying with a message that tells the device to play the dial tone file that is
stored in the flash memory of the phone. The user at phone A hears the dial tone and begins
dialing the phone number of phone B. SCCP phones send their digits to CUCM as they are
pressed (digit by digit), whereas SIP phones send their dialed digits in one message (enbloc
signaling) by default. SIP phones have options that allow them to behave similarly to SCCP
phones (Keypad Markup Language [KPML] and dial rules). CUCM performs digit analysis
against the dialed digits. If a match is found, CUCM routes the call per its configuration. If
CUCM does not find a match, a reorder tone is sent to the calling party.
IP
IP
IP
IP
CUCM Overview 11
Figure 1-4 CUCM Signaling and Media Paths
CUCM signals the calling party to initiate ringback, so the user at phone A will hear the
ringback tone. CUCM also signals the call to the destination phone, which plays the
ringdown tone. Additional information is provided to the phones to indicate the calling and
called party name and number. (Phone A will show the destination device name and
number, and phone B will show the calling party name and number.)
When the user at phone B accepts the call, CUCM sends a message to the devices letting
them know the IPv4 socket (IPv4 address and port number) information in which they
should communicate for the duration of the call. The RTP media path opens directly
between the two phones.
The Cisco IP Phones require no further communication with CUCM until either phone
invokes a feature, such as call transfer, call conferencing, or call termination.
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