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 Application Presentation Session Transport Network Data Link Mnemonic Away Pizza Sausage Throw Not Do Encapsulation Data Data Data Segments Packets Frames Routers Switches Bridges TCP/UDP/ICMP IP, Logical Address MAC, Physical Address Transport Internet Application Devices Addressing TCP/IP Model

Network Access

Physical

Please

Bits

Hubs, Repeaters

Physical, Data Link, Network, Transport, Session, Presentation, Application Please Do Not Throw Sausage Pizza Away All People Seem To Need Data Processing

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 LANs (VLANs) to Subdivide Switch:

A VLAN = a Broadcast Domain = An IP Subnet A virtual division of the switch. Routers are used to interconnect VLANs 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 VLANs 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 VLANs throughout system. VLAN information is saved in switch memory VTP Client allows only the acceptance of VLANs from the server. Information is not stored in memory. VTP Transparent mode allows the creation or deletion of VLANs 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 VLANs 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 0 1 1 0 1 0 0

Answer: -76

Converting Analog Voice to Digital continued:

Codecs convert Analog voice into Digital transmissions. Different Codecs convert in different methods with more or less complexity Available Codecs: 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 (DSPs) 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

Codecs:
Codec
G.711 Internet Low Bitrate Codec (ilBC) G.729 G.726 G.729a G.728

Bandwidth Consumed
64 Kbps 15.2 Kbps 8 Kbps 32 Kbps 8 Kbps 16 Kbps

MOS
4.1 4.1 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): 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 4-6 bytes 6 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 B: 24 bytes 4 bytes 50-57 bytes 20 mSec Sample size G.729 Codec Frame-relay network (4 byte)

Call A: 30 mSec Sample size G.711 Codec Ethernet network

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:

DSPs perform the function of sampling, encoding, and
compression of all audio signals coming into the router. DSPs might be located on the routers motherboard DSPs 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 Codecs 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 Codecs G.728, G.723, G.729, G.729b and iLBC are all high complexity

Digital Signal processors:

To calculate the number of DSPs needed use the Cisco DSP
calculator http://www.cisco.com/cgi-bin/Support/DSP/dsp-

calc.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 Audio Payload

Payload Type

Sequence Number

Time Stamp

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 VPNs 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
Automatic digit strip (POTS dial peers) Voice translation profiles

Applied 2nd

Applied 3rd

Applied 4th

Prefix digits

Applied 5th

forward-digits

End of Chapter 6+7