to a traditional telephony device, such as an
analog telephone. The gateways themselves must have IP
connectivity. In Figure , the first router has these
configuration settings: - Name: R1
- IP
address: 10.1.1.1/24
- IP interface:
FastEthernet 0/0
- Voice port: 1/0/0
- Extension of the telephone connected to the voice
port: 1111
The second router is configured with
similar settings: - Name: R2
- IP
address: 10.2.2.2/24
- IP interface:
FastEthernet 0/0
- Voice port: 1/0/0
- Extension of the telephone connected to the voice
port: 2222
Based on this information, this
configuration is applied to the first router: hostname R1
interface FastEthernet 0/0
ip address 10.1.1.1
255.255.255.0
!
dial-peer voice 1 pots
destination-pattern 1111
port 1/0/0
!
dial-peer
voice 2 voip
destination-pattern 2222
session target
ipv4:10.2.2.2
! The second router has these configuration
commands: hostname R2
interface FastEthernet 0/0
ip
address 10.2.2.2 255.255.255.0
!
dial-peer voice 1
pots
destination-pattern 2222
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1111
session
target ipv4:10.1.1.1
! The voice-specific commands in the
configurations (two dial peers in each configuration) are
highlighted in gray. A dial peer describes where to find a
telephone number, and the collection of all dial peers makes up
the call routing table of a voice gateway. Two types of dial
peers are shown in this example: POTS dial peers and VoIP dial
peers. POTS dial peers indicate that the telephone number that
is specified in the dial peer is found at a physical port. A
VoIP dial peer refers to the IP address of a VoIP device.
Figures and list the commands used for dial peers. The
Voice-Specific Commands table provides details.
Voice-Specific Commands Command
Description dial-peer voice tag type Use
the dial-peer voice command to enter the dial peer
subconfiguration mode. The tag value is a number that must be
unique for all dial peers within the same gateway. The
type value indicates the type of the dial peer (for
example, POTS or VoIP). destination-pattern
telephone_number The destination-pattern command,
entered in dial peer subconfiguration mode, defines the
telephone number that applies to the dial peer. A call that is
placed to this number is routed according to the configuration
type and port (in the case of a POTS type dial peer) or session
target (in the case of a VoIP type dial peer) of the dial
peer. port port-number The port command,
entered in POTS dial peer subconfiguration mode, defines the
port number that applies to the dial peer. Calls that
are routed using this dial peer are sent to the specified port.
The port command can be configured only on a POTS dial peer.
session target ipv4:ip-address The session
target command, entered in VoIP dial peer subconfiguration
mode, defines the IP address of the target VoIP device that
applies to the dial peer. Calls that are routed using this dial
peer are sent to the specified IP address. The session
target command can be configured only on a VoIP dial peer.
Content 2.6 VoIP Lab Exercises
2.6.1 Lab 2.1 Configure CME using the CLI and Cisco IP
Communicator Lab Activity Lab Exercise: Lab 2.1
Configure CME using the CLI and Cisco IP Communicator In this
lab, you will configure Cisco Unified Call Manager Express
using the IOS command line. On the two hosts, you will install
Cisco IP Communicator and have one host call the other. Cisco
IP Communicator is a software telephony application to simulate
a Cisco IP Phone on the desktop of a PC running Microsoft
Windows. This lab uses Cisco’s newest version of Cisco Unified
Call Manager Express at the time of this writing (CME 4.0(2))
which was tested using Cisco IOS Release 12.4(9)T1 running on a
Cisco 2800 Series router. The IP Voice image is required in
order to be able to manipulate codecs.
Content
Summary This module describes VoIP networks and the
different components these networks use in order to be able to
reliably enable voice calls. Converting analog voice signals to
digital format and digital signals back to analog format is
explained, along with the DSP modules that can be used to
perform conversion by sampling, quantization, and encoding. The
option of compression is also explained. Voice transport and
voice encapsulation are described, and the module lists
formulas for calculating the total bandwidth required for VoIP
calls. Total bandwidth depends on the different codecs that are
used. The module explains the different available IP telephony
deployment models in terms of enterprise voice implementations,
Cisco router voice gateway functions, Cisco Unified CallManager
functions, and Call Admission Control.