• CHAPTER 6 + 7• Routing Protocols• VoIP: An In-Depth Analysis

OSI Reference ModelOpen 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     Devices     Addressing     TCP/IP Model  

                       

 

Application     Away     Data            

Application

 

 

Presentation     Pizza     Data            

 

 

Session     Sausage     Data            

 

 

Transport     Throw     Segments         TCP/UDP/ICMP     Transport 

 

Network     Not     Packets     Routers    IP, Logical Address

    Internet 

 

Data Link     Do     Frames     Switches Bridges    MAC, Physical

Address   

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 commonIPv4 (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 Inter-Switch 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 serverSwitch(config)#vtp domain PHONE_NETWORKSwitch(config)#vtp password VOICEPA55Switch(config)#endSwitch#

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 10Switch(config-vlan)#name DATASwitch(config-vlan)#vlan 50Switch(config-vlan)#name VOICE Switch(config-vlan)#exitSwitch(config)#int fa0/1Switch(config-if)#switchport trunk encap dot1qSwitch(config-if)#switchport mode trunkSwitch(config-if)#switchport trunk native vlan 1Switch(config-if)#int fa0/2Switch(config-if)#switchport mode accessSwitch(config-if)#switchport access vlan 10Switch(config-if)#switchport voice vlan 50Switch(config-if)#endSwitch#

Creating Trunk Ports on a Router:

Router(config)#int fa0/0Router(config-if)#no shutRouter(config-if)#int fa0/0.10Router(config-subif)#encapsulation dot1q 10Router(config-subif)#ip address 1.10.0.1 255.255.255.0Router(config-subif)#ip helper-address 172.16.2.5Router(config-subif)#int fa0/0.50Router(config-subif)#encapsulation dot1q 50Router(config-subif)#ip address 1.50.0.1 255.255.255.0Router(config-subif)#ip helper-address 172.16.2.5Router(config-subif)#int fa0/0.1Router(config-subif)#encapsulation dot1q 1 native Router(config-subif)#ip address 1.1.0.1 255.255.255.0 Router(config-subif)#endRouter#

Creating Dynamic Host Control Protocol (DHCP) on a Router:

Router(config)#ip dhcp pool DATARouter(dhcp-config)#network 1.10.0.0 255.255.255.0Router(dhcp-config)#default-router 1.10.0.1Router(dhcp-config)#dns-server 4.2.2.2Router(dhcp-config)#ip dhcp pool VOICERouter(dhcp-config)#network 1.50.0.0 255.255.255.0Router(dhcp-config)#default-router 1.50.0.1Router(dhcp-config)#dns-server 4.2.2.2Router(dhcp-config)#option 150 ip 1.50.0.1Router(dhcp-config)#exitRouter(config)#ip dhcp excluded-address 1.10.0.1 Router(config)#ip dhcp excluded-address 1.50.0.1Router(config)#endRouter#

IP Phone Boot:

1. IP Phone connects to switchport2. Switchport senses and supplies PoE3. Via CDP phone receives voice VLAN information4. Phone sends DHCP request on voice VLAN and receives IP address, Mask and default-Gateway5. Once addressed the phone contacts TFTP server (Option 150) and downloads configuration files6. 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 64.209.210.20Router(config)#clock timezone WARWICK -5Router(config)#clock summer-time EST recurring 2 Sunday March 02:00 1 Sunday November 02:00Router(config)#endRouter#

Network Time Protocol (NTP) continued:

Router#show ntp associationsRouter# show clock

Network Delay:

1.Propagation Delay2.Handling Delay3.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

Amplitude-Modulation (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)

• Answer: -76

1 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:

Codec BandwidthMOS

Consumed

G.711 64 Kbps 4.1 Internet Low 15.2 Kbps 4.1

Bitrate Codec (ilBC) G.729 8 Kbps 3.92 G.726 32 Kbps 3.85 G.729a 8 Kbps 3.7 G.728 16 Kbps 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: 20 bytes Frame-Relay: 4-6 bytes Point-to-point Protocol (PPP): 6 bytes

Layer 3 and 4, network and transport IP: 20 bytes UDP: 8 bytes Real-time Transport Protocol (RTP): 12 bytesTypically layers 3 and 4 are always 40 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: 24 bytes MPLS: 4 bytes Ipsec: 50-57 bytes

• Call A: Call B:30 mSec Sample size 20 mSec Sample sizeG.711 Codec G.729 CodecEthernet network Frame-relay network (4

byte)

Calculating Total Bandwidth Needed per Call continued:

• Call A:

(.03 * 64Kbps) = 1.92Kbps / 8 = 240 bytes240 + 20 (ethernet) + 40 (layer 3 and 4) = 300 bytes300 * (1 / .03) = 10K bytes per second10K * 8 = 80Kbps

• Call B:

(.02 * 8Kbps) = 160bps / 8 = 20 bytes20 + 4 (frame-relay) + 40 (layer 3 and 4) = 64 bytes64 * (1 / .02) = 3.2K bytes per second3.2K * 8 = 25.6Kbps

Calculating Total Bandwidth Needed per Call Compared continued:

• Call B: G.729

(.02 * 8Kbps) = 160bps / 8 = 20 bytes20 + 4 (frame-relay) + 40 (layer 3 and 4) = 64 bytes64 * (1 / .02) = 3.2K bytes per second3.2K * 8 = 25.6Kbps

• Call B: G.711

(.02 * 64Kbps) = 128Kbps / 8 = 160 bytes160 + 4 (frame-relay) + 40 (layer 3 and 4) = 204 bytes204 * (1 / .02) = 10.2K bytes per second10.2K * 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/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 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 Leg 1: The incoming POTS call leg from x1101 on CME_A Call Leg 2: The outgoing VoIP call leg from CME_A to ROUTER_B Call Leg 3: The incoming VoIP call leg on ROUTER_B from CME_A Call Leg 4: The outgoing POTS call leg to x2510 from ROUTER_B

Configuring POTS Dial Peers:

CME_A(config)#dial-peer voice 1101 potsCME_A(config-dial-peer)#destination-pattern 1101CME_A(config-dial-peer)#port 0/0/0CME_A(config-dial-peer)#exitCME_A(config)#dial-peer voice 1102 potsCME_A(config-dial-peer)#destination-pattern 1102CME_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 potsRouter_B(config-dial-peer)#destination-pattern 2…Router_B(config-dial-peer)#no digit-stripRouter_B(config-dial-peer)#port 1/0:23

Configuring VoIP Dial Peer:

CME_A(config)#dial-peer voice 2000 voipCME_A(config-dial-peer)#destination-pattern 2…CME_A(config-dial-peer)#session target ipv4:10.1.1.2CME_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 voipRouter_B(config-dial-peer)#destination-pattern 1…Router_B(config-dial-peer)#session target ipv4:10.1.1.1Router_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]… Matches: 5551…, 5552…, 5553… (Where … is

any three digits) [14-6]555 Matches 1555, 4555, 5555, 6555 55[59]12 Matches 55512, 55912 [^1-7]..[135] 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]…… Used for 7-digit dialing• [2-9]..[2-9]…… Used for 10-digit dialing• 1[2-9]..[2-9]…… Used for 11-digit dialing• [469]11 Used for service numbers• 011T Used for international dialing

North American Dial Plan:

Router(config)#dial-peer voice 90 potsRouter(config-dial-peer)#description Service DialingRouter(config-dial-peer)#destination-pattern 9[469]11Router(config-dial-peer)#forward-digits 3Router(config-dial-peer)#port 1/0:1Router(config-dial-peer)#dial-peer voice 91 potsRouter(config-dial-peer)#description 10-Digit DialingRouter(config-dial-peer)#destination-pattern 9[2-9]..[2-9]……Router(config-dial-peer)#forward-digits 10Router(config-dial-peer)#port 1/0:1Router(config-dial-peer)#dial-peer voice 92 potsRouter(config-dial-peer)#description 11-Digit DialingRouter(config-dial-peer)#destination-pattern 91[2-9]..[2-9]……Router(config-dial-peer)#forward-digits 11Router(config-dial-peer)#port 1/0:1Router(config-dial-peer)#dial-peer voice 91 potsRouter(config-dial-peer)#description International DialingRouter(config-dial-peer)#destination-pattern 9011TRouter(config-dial-peer)#prefix 011Router(config-dial-peer)#port 1/0:1

Private Line Automatic Ringdown (PLAR):

Router(config)#voice-port 2/0/0Router(config-voiceport)#connection plar 1500Router(config-voiceport)#voice-port 2/0/1Router(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 voipRouter(config-dial-peer)#destination-pattern 555[1-3]…Router(config-dial-peer)#session target ipv4:10.1.1.1Router(config-dial-peer)#dial-peer voice 2 voipRouter(config-dial-peer)#destination-pattern 5551…Router(config-dial-peer)#session target ipv4:10.1.1.2Router(config-dial-peer)#dial-peer voice 3 voipRouter(config-dial-peer)#destination-pattern 5551Router(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 peer2. Match the called ID information (ANI) using the answer-address dial-peer configuration3. Match the caller ID information (ANI) using the destination-pattern dial-peer configuration4. Match an incoming POTS dial peer by using the port dial-peer configuration5. 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 5321Example: 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 voipArizona(config-dial-peer)#destination-pattern 6…Arizona(config-dial-peer)#session target ipv4:10.1.1.2Arizona(config-dial-peer)#preference 0Arizona(config-dial-peer)#dial-peer voice 11 potsArizona(config-dial-peer)#destination pattern 6…Arizona(config-dial-peer)#port 1/0:1Arizona(config-dial-peer)#preference 1Arizona(config-dial-peer)#no digit-stripArizona(config-dial-peer)#prefix 1512555

Texas(config)#dial-peer voice 10 voipTexas(config-dial-peer)#destination-pattern 5…Texas(config-dial-peer)#session target ipv4:10.1.1.1Texas(config-dial-peer)#preference 0Texas(config-dial-peer)#dial-peer voice 11 potsTexas(config-dial-peer)#destination pattern 5…Texas(config-dial-peer)#port 1/0:1Texas(config-dial-peer)#preference 1Texas(config-dial-peer)#no digit-stripTexas(config-dial-peer)#prefix 1480555

Using num-exp to Transform numbers:

Router(config)#voice-port 1/0/1Router(config-voiceport)#connection plar 0Router(config-voiceport)#exitRouter(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 potsRemote_RTR(config-dial-peer)#destination-pattern 911Remote_RTR(config-dial-peer)#port 1/0/0Remote_RTR(config-dial-peer)#no digit stripRemote_RTR(config-dial-peer)#dial-peer voice 11 potsRemote_RTR(config-dial-peer)#destination pattern 9911Remote_RTR(config-dial-peer)#port 1/0/0Remote_RTR(config-dial-peer)#forward-digits 3 Remote_RTR(config-dial-peer)#dial-peer voice 12 potsRemote_RTR(config-dial-peer)#destination pattern 911Remote_RTR(config-dial-peer)#port 1/0/1Remote_RTR(config-dial-peer)#no digit-strip Remote_RTR(config-dial-peer)#dial-peer voice 13 potsRemote_RTR(config-dial-peer)#destination pattern 9911Remote_RTR(config-dial-peer)#port 1/0/1Remote_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 1Router(config-translation-rule)#rule 1 /6/ /5/Router(config-translation-rule)#voice translation-profile CHANGE_DIDRouter(config-translation-profile)#translate called 1Router(config-translation-profile)#dial-peer voice 100 potsRouter(config-dial-peer)#translation-profile incoming CHANGE_DID

Translation Profile:

Translation Order:

num-exp

Automatic digit strip(POTS dial peers)

Voice translation profiles

Prefix digits

forward-digits

Applied 1st

Applied 2nd

Applied 3rd

Applied 4th

Applied 5th

End of Chapter 6+7