Information Technology Reference
In-Depth Information
transfer, conference, and so on) are still managed using SCCP or SIP, so those would not
be available until the CME router came back online. Likewise, after the users disconnect
from the call, the phone would not be available to place or receive any calls until the CME
router returned online.
Note:
This discussion assumes a CME deployment where there is no backup call-man-
agement device. This is common for smaller organizations.
Let's broaden the discussion by following a call from a Cisco IP Phone to an analog phone
attached to the PSTN shown in Figure 2-2.
PSTN
CME Router
Digital or
Analog Audio
SCCP or SIP
Analog PSTN
Phone
RTP Audio
V
Switch
Cisco IP Phone
Figure 2-2
CME Call Flow for Calls to the PSTN
As the user of the Cisco IP Phone picks up the phone to place the call, all the communica-
tion is handled by SCCP or SIP. After the user finishes dialing the phone number of the
PSTN phone, the CME router realizes (due to its dial-plan) that the call needs to exit out a
PSTN-connected interface. CME now assumes the role of voice gateway and signals to the
PSTN to establish the call on behalf of the Cisco IP Phone. Keep in mind that this is the
“old school” signaling discussed in Chapter 1 since the CME router is attached to the
PSTN using a digital (T1/E1) or analog (FXO) trunk. Once the audio for the call connects,
the CME router assumes the role of converting between VoIP audio and PSTN audio. Be-
cause this conversion is processor intensive, the CME router is equipped with Digital Sig-
nal Processors (DSP), which are simply additional “mini processors” dedicated to voice
functions.
