VOICE OVER IP FUNDAMENTALS
•
•
•
CHAPTER 6 + 7
Routing Protocols
VoIP: An In-Depth Analysis
OSI Reference Model
Open System Interconnection Model
• Seven Layered Model
• Developed by the International Standards Organization
• Predated by the TCP/IP Model
OSI / TCP/IP Model
OSI Model
Mnemonic
Encapsulation
Application
Away
Data
Presentation
Pizza
Data
Session
Sausage
Data
Transport
Throw
Segments
Network
Not
Packets
Data Link
Do
Frames
Physical
Please
Bits
Devices
Addressing
TCP/IP Model
Application
TCP/UDP/ICMP
Transport
Routers
IP, Logical
Address
Internet
Switches Bridges
MAC, Physical
Address
Hubs, Repeaters
Physical, Data Link, Network, Transport, Session, Presentation, Application
Please Do Not Throw Sausage Pizza Away
All People Seem To Need Data Processing
Network Access
OSI Layers:
• Application
 Provides Services to applications
 E-Mail
 Web Browsing
 Word Processing
• Presentation
 Formats Data
 Encryption
 Compression
 ASCII, EBCDIC
OSI Layers:
• Session
 Establishes, Manages and Terminates Sessions
between applications
 Dialog Control
• Transport
 Ensures Reliable Transport of Data
 Transmission Control Protocol (TCP)
 User Datagram Protocol (UDP)
 Reliable Transport Protocol (RTP)
 Port Numbers
OSI Layers:
• Network
 Packet Formatting
 Logical Addressing
 Routing
• Data Link
 Provides reliable transport across a physical link
 Physical Addressing
 Media Access Control (MAC)
OSI Layers:
• Physical
 Converting data to physical impulses
 EIA/TIA-232
 V.35
 RS-449
 802.3
 Others
Addressing:
• Physical (MAC)
 48 bit Hexadecimal Address burned into device
memory
 24 bits Organizational Unique Identifier (OUI)
 24 bits Serial Number
 Layer 2 of the OSI Model
Addressing:
•Logical (IP,IPX,AppleTalk)





IP Most common
IPv4 (32 bits)
IPv6 (128 bits)
Dotted Decimal Format (IPv4)
Classes A, B, C, Multicast, and Expirmental
 Class A: 0.0.0.0 through 127.255.255.255
 Class B: 128.0.0.0 through 191.255.255.255
 Class C: 192.0.0.0 through 223.255.255.255
 Multicast: 224.0.0.0 through 239.255.255.255
 Experimental: 240.0.0.0 through 255.255.255.254
Connecting to the Network:
Routing Protocols:
• Distance-Vector Routing
 View from directly connected neighbors
 BGP
 EIGRP
 RIP
• Link-state Routing
 View of entire network
 IS-IS
 OSPF
Using Virtual LAN’s (VLAN’s) to Subdivide Switch:
•
•
•
•
A VLAN = a Broadcast Domain = An IP Subnet
A virtual division of the switch.
Routers are used to interconnect VLAN’s
Benefits:
 Increased performance
 Improved manageability
 Physical topology independence
 Increased security
Switch Trunking:
• Switches can be interconnected via a single
connection
• Uses either IEEE 802.1Q (Standard) or InterSwitch Link protocol (ISL) a Cisco proprietary.
• Native VLAN carries all management information
• All frames are “Tagged” to cross the trunk link
except for the native VLAN frames.
• Tagging adds bits onto frame which are removed
prior to exiting the switch on any line not a trunk
• Tagging adds delay
• Tagging saves physical ports
• VLAN’s are distributed to all switches via Virtual
Trunking Protocol (VTP)
Virtual Trunking Protocol:
• Switches exchange VLAN information automatically
• VTP Domain Names and passwords are case sensitive
• VTP Modes are server, client or transparent
• VTP Server allows the creation or deletion of
VLAN’s throughout system. VLAN information is
saved in switch memory
• VTP Client allows only the acceptance of VLAN’s
from the server. Information is not stored in
memory.
• VTP Transparent mode allows the creation or
deletion of VLAN’s of local significance only. VLAN
information is stored in switch memory. Will pass VTP
information to other switches within the same domain.
Virtual Trunking Protocol continued:
• Each VTP change increased VTP revision number.
Highest revision number is distributed through out
system
• Configuration:
Switch(config)#vtp mode server
Switch(config)#vtp domain PHONE_NETWORK
Switch(config)#vtp password VOICEPA55
Switch(config)#end
Switch#
Virtual Trunking Protocol continued:
• Interface trunking modes:
 Dynamic desirable: Cisco default. Will become
trunk depending on mode and device attached.
 Dynamic auto: Will become a trunk depending on
mode and device attached but will not actively try
to negotiate a trunk link.
 Trunk: Will be in trunk mode but will negotiate
with either dynamic auto, dynamic desirable using
Dynamic Trunking Protocol (DTP).
 Access: Not in a trunk mode. Gives access to
one Data VLAN and one Voice VLAN only.
 Nonegotiate: Disables DTP messages on
interface
Creating VLAN’s on a Switch:
Switch(config)#vlan 10
Switch(config-vlan)#name DATA
Switch(config-vlan)#vlan 50
Switch(config-vlan)#name VOICE
Switch(config-vlan)#exit
Switch(config)#int fa0/1
Switch(config-if)#switchport trunk encap dot1q
Switch(config-if)#switchport mode trunk
Switch(config-if)#switchport trunk native vlan 1
Switch(config-if)#int fa0/2
Switch(config-if)#switchport mode access
Switch(config-if)#switchport access vlan 10
Switch(config-if)#switchport voice vlan 50
Switch(config-if)#end
Switch#
Creating Trunk Ports on a Router:
Router(config)#int fa0/0
Router(config-if)#no shut
Router(config-if)#int fa0/0.10
Router(config-subif)#encapsulation dot1q 10
Router(config-subif)#ip address 1.10.0.1 255.255.255.0
Router(config-subif)#ip helper-address 172.16.2.5
Router(config-subif)#int fa0/0.50
Router(config-subif)#encapsulation dot1q 50
Router(config-subif)#ip address 1.50.0.1 255.255.255.0
Router(config-subif)#ip helper-address 172.16.2.5
Router(config-subif)#int fa0/0.1
Router(config-subif)#encapsulation dot1q 1 native
Router(config-subif)#ip address 1.1.0.1 255.255.255.0
Router(config-subif)#end
Router#
Creating Dynamic Host Control Protocol (DHCP) on a
Router:
Router(config)#ip dhcp pool DATA
Router(dhcp-config)#network 1.10.0.0 255.255.255.0
Router(dhcp-config)#default-router 1.10.0.1
Router(dhcp-config)#dns-server 4.2.2.2
Router(dhcp-config)#ip dhcp pool VOICE
Router(dhcp-config)#network 1.50.0.0 255.255.255.0
Router(dhcp-config)#default-router 1.50.0.1
Router(dhcp-config)#dns-server 4.2.2.2
Router(dhcp-config)#option 150 ip 1.50.0.1
Router(dhcp-config)#exit
Router(config)#ip dhcp excluded-address 1.10.0.1
Router(config)#ip dhcp excluded-address 1.50.0.1
Router(config)#end
Router#
IP Phone Boot:
1.
2.
3.
4.
IP Phone connects to switchport
Switchport senses and supplies PoE
Via CDP phone receives voice VLAN information
Phone sends DHCP request on voice VLAN and
receives IP address, Mask and default-Gateway
5. Once addressed the phone contacts TFTP server
(Option 150) and downloads configuration files
6. Phone contacts first call processing center (CME
Router) and registers. If unable to contact will
contact additional centers as listed in
configuration
Network Time Protocol (NTP):
• Assigns correct date and time to voice mail
• Displays correct date and time on phone
• Synchronizes system
Router(config)#ntp server 64.209.210.20
Router(config)#clock timezone WARWICK -5
Router(config)#clock summer-time EST recurring 2
Sunday March 02:00 1 Sunday November 02:00
Router(config)#end
Router#
Network Time Protocol (NTP) continued:
Router#show ntp associations
Router# show clock
Network Delay:
1. Propagation Delay
2. Handling Delay
3. Queuing Delay
* Total acceptable delay is 150 mSec
Jitter:
 Variation in delay affecting packet arrival time
Converting Analog to Digital:
• Sample the signal
•Quantize the signal
•Encode the quantized value into binary
format:
•Optionally compress the sample to save
bandwidth.
Sample the Signal:
• How often to Sample?
 Nyquist – 18,000 Samples per second!
 Realistically to recognize voice and mood
8,000 Samples per second.
 Result less quality less bandwidth
 Process referred to as Pulse AmplitudeModulation (PAM)
Quantize the Signal:
• How many Digits?
 Known as Quantization
 Divided into sixteen (16) segments. 0
through 7 positive and 0 through 7
negative
 Values are not evenly spaced to allow
for more accurate recreation of voice
patterns
Encode the Quantized Signal:
• How many Digits?
 Each Quantized value is encoded into an
eight bit (8) binary number.
 Total bandwidth is equal to eight bits
for each sample times eight thousand
samples per second.
 8 X 8000 = 64Kbps
Compress the Sample:
• Why?
 Save bandwidth.
 Reduces quality of voice
 As low as 8Kbps
Converting Analog Voice to Digital:
•
The average human can hear frequencies of 20-20,000 Hz
• Human speech uses frequencies from 200-9000 Hz
• Telephone channels typically transmit frequencies of 300-3400 Hz
• The Nyquist theorem is able to reproduce frequencies of 300-4000 Hz
Converting Analog Voice to Digital continued:
• Sample at twice the highest frequency to reproduce accurately
(Nyquist)
• Quantization is the term used to describe the process of
converting an analog signal into a numeric quantity
• Since an eight (8) bit binary number can represent a value from
zero (0) through two-hundred fifty-five (255) we use the Most
Significant Digit (MSD) to represent positive/negative value
• A zero (0) in the MSD represents a positive (+) value
• A one (1) in the MSD represents a negative (-) value
• The result is a range of zero through positive one-hundred
twenty-seven (0 through +127) and negative one through negative
one-hundred twenty-seven (-1 through -127)
1
• Answer: -76
0
1
1
0
1
0
0
Converting Analog Voice to Digital continued:
• Codec’s convert Analog voice into Digital
transmissions.
• Different Codec’s convert in different methods with
more or less complexity
• Available Codec’s:
 G.711
 Internet low Bitrate Codec (iLBC)
 G.729
 G.726
 G.729a
 G.728
• Is the Codec supported in the system
• How many Digital Signal Processors (DSP’s) are used
Converting Analog Voice to Digital continued:
•
•
•
•
Does the Codec meet satisfactory quality levels
How much bandwidth does the Codec consume
How does the Codec handle packet loss
Does the Codec support multiple sample size
Codec’s:
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Codec
Bandwidth
Consumed
MOS
G.711
Internet Low
Bitrate Codec (ilBC)
G.729
G.726
G.729a
G.728
64 Kbps
15.2 Kbps
4.1
4.1
8 Kbps
32 Kbps
8 Kbps
16 Kbps
3.92
3.85
3.7
3.61
• MOS (Mean Opinion Score) is determined by listeners listening to the
phrase “Nowadays, a chicken leg is a rare dish.” and scoring the quality
of the connection on a one to five scale.
Calculating Total Bandwidth Needed per Call:
• Determine sample size: A larger sample is more
efficient (Example: 30 bytes of voice to 50 bytes of
overhead 30/80x100%=37.5% is Voice)(Example: 20
bytes of voice to 50 bytes of overhead
20/70x100%=28.5% is voice)
• A larger sample takes longer to prepare, so in
circuits with delay the voice call will not be as good.
• Bandwidth can be saved using Voice Activity
Detection (VAD) where no packets are sent during a
time when there is no voice
• VAD can account for 35-40% of total call time
• RTP header compression does not repeat the header
after the first packet since the information will stay
the same for the length of the call saving 40%
Calculating Total Bandwidth Needed per Call
continued:
• Determine CODEC used
• Determine sample size
• Determine layer overhead
 Layer 2 datalink
 Ethernet:
 Frame-Relay:
 Point-to-point Protocol (PPP):
20 bytes
4-6 bytes
6 bytes
 Layer 3 and 4, network and transport
 IP:
 UDP:
 Real-time Transport Protocol (RTP):
Typically layers 3 and 4 are always 40 bytes
20 bytes
8 bytes
12 bytes
Calculating Total Bandwidth Needed per Call
continued:
•
Bytes-per-packet = (Sample_size * Codec_bandwidth) / 8
• Total_bandwidth = Packet_size * Packets_per_second
• Add



any additional overhead:
GRE/L2TP:
MPLS:
Ipsec:
• Call A:
30 mSec Sample size
G.711 Codec
Ethernet network
24 bytes
4 bytes
50-57 bytes
Call B:
20 mSec Sample size
G.729 Codec
Frame-relay network (4 byte)
Calculating Total Bandwidth Needed per Call
continued:
• Call A:
(.03 * 64Kbps) = 1.92Kbps / 8 = 240 bytes
240 + 20 (ethernet) + 40 (layer 3 and 4) = 300 bytes
300 * (1 / .03) = 10K bytes per second
10K * 8 = 80Kbps
•
Call B:
(.02 * 8Kbps) = 160bps / 8 = 20 bytes
20 + 4 (frame-relay) + 40 (layer 3 and 4) = 64 bytes
64 * (1 / .02) = 3.2K bytes per second
3.2K * 8 = 25.6Kbps
Calculating Total Bandwidth Needed per Call
Compared continued:
• Call B: G.729
(.02 * 8Kbps) = 160bps / 8 = 20 bytes
20 + 4 (frame-relay) + 40 (layer 3 and 4) = 64 bytes
64 * (1 / .02) = 3.2K bytes per second
3.2K * 8 = 25.6Kbps
•
Call B: G.711
(.02 * 64Kbps) = 128Kbps / 8 = 160 bytes
160 + 4 (frame-relay) + 40 (layer 3 and 4) = 204 bytes
204 * (1 / .02) = 10.2K bytes per second
10.2K * 8 = 81.6Kbps
• Savings of 68.6% using the G.729 Codec!
Digital Signal processors:
• DSP’s perform the function of sampling, encoding, and
compression of all audio signals coming into the router.
• DSP’s might be located on the routers motherboard
• DSP’s might also be add on modules similar to SIMM memory
modules on the motherboard called Packet Voice DSP Modules
(PVDM)
• DSP modules can contain multiple DSP circuits
 PVDM2-8: Provides .5 DSP chip
 PVDM2-16: Provides 1 DSP chip
 PVDM2-32: Provides 2 DSP chips
 PVDM2-48: Provides 3 DSP chips
 PVDM2-64: Provides 4 DSP chips
• Codec’s G.711 (a-law and u-law) (u-law is United States,
Japan) (a-law All others), G.726, G.729a, and G.729ab are all
of medium complexity
• Codec’s G.728, G.723, G.729, G.729b and iLBC are all high
complexity
Digital Signal processors:
• To calculate the number of DSP’s needed use the Cisco DSP
calculator http://www.cisco.com/cgi-bin/Support/DSP/dspcalc.pl (Must have Cisco CCO account)
RTP and RTCP:
• Real-time Transport Protocol (RTP) operates at the transport
layer (layer 4) of the OSI model
• Real-time Transport Control Protocol (RTCP) also operates at
the transport layer (layer 4) of the OSI model
• They both work on top of User datagram Protocol (UDP)
• Two transport layer protocols simultaneously working is highly
unusual but is what happens with voice and video!
• UDP works as normal to provide port numbers and header
checksums
• RTP adds time stamps, sequence numbers, and header
information
Data Link
IP
RTP
UDP
Payload
Type
Sequence
Number
Time
Stamp
Audio
Payload
RTP and RTCP continued:
• The payload will specify if the packet is handling voice or
video
• Once established RTP will use even numbered port from
between 16,384 and 32,767
• RTP streams are one-way! If a two-way communication takes
place then a second session is established
• RTCP also engages at the same time and establishes a session
using an odd numbered port from the same range that follows
the RTC even numbered port chosen
• RTCP will account for:
 Packet Count
 Packet Delay
 Packet Loss
 Jitter (delay variations)
• RTP carries the voice while RTCP does the accounting
• RTCP is used to evaluate if there is enough bandwidth or
services to complete a call of good quality
Internet Low Bitrate Codec (iLBC):
• Industry nonproprietary compression codec that is universally
supported
• Developed in 2000 to provide high-quality, bandwidth-savvy,
available to all industry vendors
• Provides a bit rate of 15.2 Kbps when coded using a 20 mSec
sample size, and 13.3 Kbps when using a 30 mSec sample size
• Is a high complexity codec (more DSP required)
• High quality approaching G.711 (64 Kbps). The best of any
compression codec
• Limited support at this time. Cisco phone models that support
iLBC: 7906G, 7911G, 7921G, 7942G, 7945G, 7962G, 7965G,
and 7975G
Speech Quality, Echo:
• Impedance Mismatch
Speech Quality:
• Packet Loss
Speech Quality:
• Voice Activity Detection (VAD)
Dial Plan:
• Plans for growth
• Cost of leased circuits or VPN’s
• Cost of additional equipment for packet voice
• Number overlap (When one or more sites have the
same phone numbers)
• Call-flows (The call patterns from each side)
• Busy hour (The time of day when the highest
number of calls are offered on a circuit)
Configuring Dial Peers:
• POTS dial peer: Used to define voice reachability
information for any traditional (analog) connection
• VoIP dial peer: Used to define any voice connection
available through IP addressing
Call Legs:
• Any voice connection too or from a voice port or
connection or voice device




Call
Call
Call
Call
Leg
Leg
Leg
Leg
1:
2:
3:
4:
The
The
The
The
incoming
outgoing
incoming
outgoing
POTS call leg from x1101 on CME_A
VoIP call leg from CME_A to ROUTER_B
VoIP call leg on ROUTER_B from CME_A
POTS call leg to x2510 from ROUTER_B
Configuring POTS Dial Peers:
CME_A(config)#dial-peer voice 1101 pots
CME_A(config-dial-peer)#destination-pattern 1101
CME_A(config-dial-peer)#port 0/0/0
CME_A(config-dial-peer)#exit
CME_A(config)#dial-peer voice 1102 pots
CME_A(config-dial-peer)#destination-pattern 1102
CME_A(config-dial-peer)#port 0/0/1
Configuring Dial Peers:
• Router#show dial-peer voice summary
Configuring POTS Dial Peer for T1:
Router_B(config)#dial-peer voice 2000 pots
Router_B(config-dial-peer)#destination-pattern 2…
Router_B(config-dial-peer)#no digit-strip
Router_B(config-dial-peer)#port 1/0:23
Configuring VoIP Dial Peer:
CME_A(config)#dial-peer voice 2000 voip
CME_A(config-dial-peer)#destination-pattern 2…
CME_A(config-dial-peer)#session target ipv4:10.1.1.2
CME_A(config-dial-peer)#codec g711ulaw
• If the configured codec does not match the opposite end then the call
will fail. The default codec is G.729
Router_B(config)#dial-peer voice 1000 voip
Router_B(config-dial-peer)#destination-pattern 1…
Router_B(config-dial-peer)#session target ipv4:10.1.1.1
Router_B(config-dial-peer)#codec g711ulaw
Using Dial-Peer Wildcards:
• Period (.): Will match any digit
• Plus(+): matches one or more instances of the preceding digits
• Brackets ([]): Matches a range of digits
• T: matches any dialed number from 0-32 digits
• Carrot (^): Does not match
• Comma (,): Inserts a one-second pause between dialed digits
• Example:
 555[1-3]…
 [14-6]555
 55[59]12
 [^1-7]..[135]
Matches: 5551…, 5552…, 5553… (Where … is
any three digits)
Matches 1555, 4555, 5555, 6555
Matches 55512, 55912
Matches 8..1, 8..3, 8..5, 9..1, 9..3, 9..5
(Where is any two digits)
Digit Manipulation:
Digit Manipulation Problem:
Digit Manipulation Problem Answer:
North American Dial Plan:
• [2-9]……
• [2-9]..[2-9]……
• 1[2-9]..[2-9]……
• [469]11
• 011T
Used
Used
Used
Used
Used
for
for
for
for
for
7-digit dialing
10-digit dialing
11-digit dialing
service numbers
international dialing
North American Dial Plan:
Router(config)#dial-peer voice 90 pots
Router(config-dial-peer)#description Service Dialing
Router(config-dial-peer)#destination-pattern 9[469]11
Router(config-dial-peer)#forward-digits 3
Router(config-dial-peer)#port 1/0:1
Router(config-dial-peer)#dial-peer voice 91 pots
Router(config-dial-peer)#description 10-Digit Dialing
Router(config-dial-peer)#destination-pattern 9[2-9]..[2-9]……
Router(config-dial-peer)#forward-digits 10
Router(config-dial-peer)#port 1/0:1
Router(config-dial-peer)#dial-peer voice 92 pots
Router(config-dial-peer)#description 11-Digit Dialing
Router(config-dial-peer)#destination-pattern 91[2-9]..[2-9]……
Router(config-dial-peer)#forward-digits 11
Router(config-dial-peer)#port 1/0:1
Router(config-dial-peer)#dial-peer voice 91 pots
Router(config-dial-peer)#description International Dialing
Router(config-dial-peer)#destination-pattern 9011T
Router(config-dial-peer)#prefix 011
Router(config-dial-peer)#port 1/0:1
Private Line Automatic Ringdown (PLAR):
Router(config)#voice-port 2/0/0
Router(config-voiceport)#connection plar 1500
Router(config-voiceport)#voice-port 2/0/1
Router(config-voiceport)#connection plar 1500
Call Processing:
• Most specific pattern wins
• Once a match is found the call is processed
Router(config)#dial-peer voice 1 voip
Router(config-dial-peer)#destination-pattern 555[1-3]…
Router(config-dial-peer)#session target ipv4:10.1.1.1
Router(config-dial-peer)#dial-peer voice 2 voip
Router(config-dial-peer)#destination-pattern 5551…
Router(config-dial-peer)#session target ipv4:10.1.1.2
Router(config-dial-peer)#dial-peer voice 3 voip
Router(config-dial-peer)#destination-pattern 5551
Router(config-dial-peer)#session target ipv4:10.1.1.3
If a user dials 5551234 dial-peer 3 will be used because it is a more
specific match. Router will drop the last three digits and only route the
5551 (Useful for emergency calls)
Matching Inbound and Outbound Dial Peers:
1. Match the dialed number (DNIS) using the incoming called
number dial peer
2. Match the called ID information (ANI) using the answeraddress dial-peer configuration
3. Match the caller ID information (ANI) using the destinationpattern dial-peer configuration
4. Match an incoming POTS dial peer by using the port dial-peer
configuration
5. If no match has been found using the previous four methods,
use dial peer 0
Dial Peer 0:
• Default Dial Peer
 Uses any voice codec (Not hard coded)
 No DTMF relay: DTMF relay sends dial tones outside of
the audio stream
 IP Precedence 0: Strips all QoS markings. Calls will now
be sent as if they were normal data
 Voice Activity Detection (VAD) enabled: Allows bandwidth
savings by not transmitting dead time
 No Resource Reservation Protocol (RSVP) support: The
router will not reserve end-to-end bandwidth
 Fax-rate voice: The router will limit fax bandwidth to
that of the VoIP codec. Can devastate fax calls
 No application support: calls cannot be referred to outside
applications
 No Direct Inward Dial (DID) support: Cannot use the DID
feature to forward calls to an internal device from an PSTN
source
Digit Manipulation:
• prefix digits: Allows for digits to be added to be specified
• forward-digits number: Allows for the number of digits that will
be forwarded
• [no] digit-strip: Enables (default) or disables digit stripping
• num-exp: Transforms any number dialed that matches pattern.
Example: num-exp 4… 5…
Call 4321 converted to 5321
Example: num-exp 0 5000
Call 0 converted to 5000
• voice translation profile: Allows a translation profile of up to 15
rules to be transform the number
POTS Failover:
• If the VoIP network fails, the phone system should
automatically switch to the POTS system
POTS Failover Configuration:
Arizona(config)#dial-peer voice 10 voip
Arizona(config-dial-peer)#destination-pattern 6…
Arizona(config-dial-peer)#session target ipv4:10.1.1.2
Arizona(config-dial-peer)#preference 0
Arizona(config-dial-peer)#dial-peer voice 11 pots
Arizona(config-dial-peer)#destination pattern 6…
Arizona(config-dial-peer)#port 1/0:1
Arizona(config-dial-peer)#preference 1
Arizona(config-dial-peer)#no digit-strip
Arizona(config-dial-peer)#prefix 1512555
Texas(config)#dial-peer voice 10 voip
Texas(config-dial-peer)#destination-pattern 5…
Texas(config-dial-peer)#session target ipv4:10.1.1.1
Texas(config-dial-peer)#preference 0
Texas(config-dial-peer)#dial-peer voice 11 pots
Texas(config-dial-peer)#destination pattern 5…
Texas(config-dial-peer)#port 1/0:1
Texas(config-dial-peer)#preference 1
Texas(config-dial-peer)#no digit-strip
Texas(config-dial-peer)#prefix 1480555
Using num-exp to Transform numbers:
Router(config)#voice-port 1/0/1
Router(config-voiceport)#connection plar 0
Router(config-voiceport)#exit
Router(config)#num-exp 0 5000
• Connects any dialed 0, to the receptionist within the company at
extension 5000
POTS Lines for Emergency Calls:
Remote_RTR(config)#dial-peer voice 10 pots
Remote_RTR(config-dial-peer)#destination-pattern 911
Remote_RTR(config-dial-peer)#port 1/0/0
Remote_RTR(config-dial-peer)#no digit strip
Remote_RTR(config-dial-peer)#dial-peer voice 11 pots
Remote_RTR(config-dial-peer)#destination pattern 9911
Remote_RTR(config-dial-peer)#port 1/0/0
Remote_RTR(config-dial-peer)#forward-digits 3
Remote_RTR(config-dial-peer)#dial-peer voice 12 pots
Remote_RTR(config-dial-peer)#destination pattern 911
Remote_RTR(config-dial-peer)#port 1/0/1
Remote_RTR(config-dial-peer)#no digit-strip
Remote_RTR(config-dial-peer)#dial-peer voice 13 pots
Remote_RTR(config-dial-peer)#destination pattern 9911
Remote_RTR(config-dial-peer)#port 1/0/1
Remote_RTR(config-dial-peer)#forward-digits 3
Translation Profile:
• Define the rules that dictate how the router will transform
the number
• Associate the rules to a profile
• Associate the profile to a dial peer
Router(config)#voice translation-rule 1
Router(config-translation-rule)#rule 1 /6/ /5/
Router(config-translation-rule)#voice translation-profile CHANGE_DID
Router(config-translation-profile)#translate called 1
Router(config-translation-profile)#dial-peer voice 100 pots
Router(config-dial-peer)#translation-profile incoming CHANGE_DID
Translation Profile:
Translation Order:
Applied 1st
num-exp
Applied 2nd
Automatic digit strip
(POTS dial peers)
Applied 3rd
Voice translation profiles
Applied 4th
Prefix digits
Applied 5th
forward-digits
End of Chapter 6+7