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Chrysler 2007 Town and Country Automobile User Manual

1. Configure the dialer interface or virtual template as defined in the relevant chapters of the Dial Solutions Configuration Guide Configure Multilink PPP and interleaving on the interface or template To configure Multilink PPP and interleaving on a configured and operational interface or virtual interface template use the following commands in interface mode Step Command Purpose 1 ppp multilink Enable Multilink PPP 2 ppp multilink interleave Enable real time packet interleaving 3 ppp multilink fragment delay milliseconds Optionally configure a maximum fragment delay 4 ip rtp reserve lowest UDP port Reserve a special queue for real time packet flows range of ports maximum bandwidth to specified destination User Datagram Protocol UDP ports allowing real time traffic to have higher priority than other flows This is only applicable if you have not configured RSVP Note The ip rtp reserve command can be used instead of configuring RSVP If you configure RSVP this command is not required Configuring Voice over IP for the Cisco 3600 Series VC 19 Configure IP Networks for Real Time Voice Traffic For more information about Multilink PPP refer to the Configuring Media Independent PPP and Multilink PPP chapter in the Dial Solutions Configuration Guide Multilink PPP Configuration Example The following example defines a virtual interface template that enables Multilink PPP with interlea
2. Configuring Voice over IP for the Cisco 3600 Series This chapter shows you how to configure Voice over IP VoIP on the Cisco 3600 series For a description of the commands used to configure Voice over IP refer to the Voice Related Commands chapter in the Voice Video and Home Applications Command Reference VoIP enables a Cisco 3600 series router to carry voice traffic for example telephone calls and faxes over an IP network Voice over IP is primarily a software feature however to use this feature on a Cisco 3600 series router you must install a voice network module VNM The VNM can hold either two or four voice interface cards VICs each of which is specific to a particular signaling type associated with a voice port For more information about the physical characteristics installing or configuring a VNM in your Cisco 3600 series router refer to the Voice Network Module and Voice Interface Card Configuration Note that came with your VNM Voice over IP offers the following benefits Toll bypass Remote PBX presence over WANs Unified voice data trunking POTS Internet telephony gateways How Voice over IP Processes a Telephone Call Before configuring Voice over IP on your Cisco 3600 series router it helps to understand what happens at an application level when you place a call using Voice over IP The general flow of a two party voice call using Voice over IP is as follows 1 The user picks up the ha
3. configuration you can try to resolve the problem by performing the following tasks Ping the associated IP address to confirm connectivity If you cannot successfully ping your destination refer to the Network Protocols Configuration Guide Part 1 Use the show dial peer voice command to verify that the operational status of the dial peer is up Use the show dialplan number command on the local and remote routers to verify that the data is configured correctly on both If you have configured number expansion use the show num exp command to check that the partial number on the local router maps to the correct full E 164 telephone number on the remote router If you have configured a CODEC value there can be a problem if both VoIP dial peers on either side of the connection have incompatible CODEC values Make sure that both VoIP peers have been configured with the same CODEC value Use the debug vpm spi command to verify the output string the router dials is correct Use the debug cch323 rtp command to check RTP packet transport Use the debug cch323 h225 command to check the call setup Optimize Dial Peer and Network Interface Configurations Depending on how you have configured your network interfaces you might need to configure additional VoIP dial peer parameters This section describes the following topics Configure IP Precedence for Dial Peers Configure RSVP for Dial Peers Configure CODEC and VAD for Dial Peers Configure IP
4. 16 65 182 no shutdown PSTN Gateway Access Using FXO Connection PLAR Mode The following example shows how to configure Voice over IP to link users with the PSTN Gateway using an FXO connection PLAR mode In this example PSTN users in Salt Lake City Utah can dial a local number and establish a private line connection in a remote location As in the previous example Router SLC in Salt Lake City is connected directly to the PSTN through an FXO interface Figure 15 illustrates the topology of this connection example Figure 15 PSTN Gateway Access Using FXO Connection PLAR Mode IP cloud Router SJ ao Ss 1 408 555 4000 172 16 1 123 San Jose voice port 1 0 0 PLAR connection PSTN user ie Router SLC S 172 16 65 182 Voice port Salt Lake City 1 0 0 S6618 Note This example assumes that the company already has established a working IP connection between its two remote offices VC 46 Voice Video and Home Applications Configuration Guide PSTN Gateway Access Using FXO Connection PLAR Mode Configuration for Router SJ Configure pots dial peer 1 dial peer voice 1 pots destination pattern 14085554000 port 1 0 0 Configure voip dial peer 2 dial peer voice 2 voip destination pattern 9 session target ipv4 172 16 65 182 Configure the serial interface interface serial 0 0 clock rate 2000000 ip address 172 16 1 123 no shutdown Configuration for Router
5. 3600 Series VC 41 Voice over IP Configuration Examples Configure IGRP router igrp 888 network 10 0 0 network 20 0 0 network 40 0 0 0 0 0 Configuration for Router RLB 2 VC 42 hostname r1ib 2 Create pots dial peer 2 dial peer voice 2 pots Define its associated telephone number and voice port destination pattern 4155554000 port 1 0 0 Create voip dial peer 20 dial peer voice 20 voip Define its associated telephone number and IP address destination pattern 4085554000 session target ipv4 10 0 0 1 Configure serial interface 0 0 interface Serial0 0 ip address 40 0 0 1 255 0 0 0 no ip mroute cache Configure RTP header compression ip rtp header compression ip rtp compression connections 25 Enable RSVP on this interface ip rsvp bandwidth 96 96 fair queue 64 256 3 clockrate 64000 Configure IGRP router igrp 888 network 10 0 0 0 network 20 0 0 0 network 40 0 0 0 Voice Video and Home Applications Configuration Guide Linking PBX Users with E amp M Trunk Lines Linking PBX Users with E amp M Trunk Lines The following example shows how to configure Voice over IP to link PBX users with E amp M trunk lines In this example a company wants to connect two offices one in San Jose California and the other in Salt Lake City Utah Each office has an internal telephone network using PBX connected to the voice network by an E amp M interface Both the Salt Lake City and the San
6. Home Applications Configuration Guide Configure RTP Header Compression You should configure RTP header compression if the following conditions exist in your network Slow links Need to save bandwidth Note RTP header compression should not be used on links greater than 2 Mbps Perform the following tasks to configure RTP header compression for Voice over IP The first task is required the second task is optional Enable RTP Header Compression on a Serial Interface Change the Number of Header Compression Connections Enable RTP Header Compression on a Serial Interface To use RTP header compression you need to enable compression on both ends of a serial connection To enable RTP header compression use the following command in interface configuration mode Command Purpose ip rtp header compression passive Enable RTP header compression If you include the passive keyword the software compresses outgoing RTP packets only if incoming RTP packets on the same interface are compressed If you use the command without the passive keyword the software compresses all RTP traffic Change the Number of Header Compression Connections By default the software supports a total of 16 RTP header compression connections on an interface To specify a different number of RTP header compression connections use the following command in interface configuration mode Command Purpose ip rtp compression connect
7. Precedence for Dial Peers If you want to give real time voice traffic a higher priority than other network traffic you can weight the voice data traffic associated with a particular VoIP dial peer by using IP Precedence IP Precedence scales better than RSVP but provides no admission control VC 32 Voice Video and Home Applications Configuration Guide Configure RSVP for Dial Peers To give real time voice traffic precedence over other IP network traffic use the following commands beginning in global configuration mode Step Command Purpose 1 dial peer voice number voip Enter the dial peer configuration mode to configure a VolP peer 2 ip precedence number Select a precedence level for the voice traffic associated with that dial peer In IP Precedence the numbers 1 through 5 identify classes for IP flows the numbers 6 through 7 are used for network and backbone routing and updates For example to ensure that voice traffic associated with VoIP dial peer 103 is given a higher priority than other IP network traffic enter the following dial peer voice 103 voip ip precedence 5 In this example when an IP call leg is associated with VoIP dial peer 103 all packets transmitted to the IP network via this dial peer will have their precedence bits set to 5 If the networks receiving these packets have been configured to recognize precedence bits the packets will be given priority over packets with a lower configured p
8. Relay for Voice over IP section for information about deploying Voice over IP over Frame Relay 3 Configure Number Expansion Use the num exp command to configure number expansion if your telephone network is configured so that you can reach a destination by dialing only a portion an extension number of the full E 164 telephone number Refer to the Configure Number Expansion section for information about number expansion 4 Configure Dial Peers Use the dial peer voice command to define dial peers and switch to the dial peer configuration mode Each dial peer defines the characteristics associated with a call leg A call leg is a discrete segment of a call connection that lies between two points in the connection An end to end call is comprised of four call legs two from the perspective of the source access server and two from the perspective of the destination access server Dial peers are used to apply attributes to call legs and to identify call origin and destination There are two different kinds of dial peers a _POTS Dial peer describing the characteristics of a traditional telephony network connection POTS peers point to a particular voice port on a voice network device To minimally configure a POTS dial peer you need to configure the following two characteristics associated telephone number and logical interface Use the destination pattern command to associate a telephone number with a POTS peer Use the port command
9. SLC Configure pots dial peer 1 dial peer voice 1 pots destination pattern 9 port 1 0 0 Configure voip dial peer 2 dial peer voice 2 voip destination pattern 14085554000 session target ipv4 172 16 1 123 Configure the voice port voice port 1 0 0 connection plar 14085554000 Configure the serial interface interface serial 0 0 ip address 172 16 65 182 no shutdown Configuring Voice over IP for the Cisco 3600 Series VC 47 Voice over IP Configuration Examples VC 48 Voice Video and Home Applications Configuration Guide
10. VoIP DTMF Dual tone multifrequency Use of two simultaneous voice band tones for dial such as touch tone E amp M E amp M stands for recEive and transMit or Ear and Mouth E amp M is a trunking arrangement generally used for two way switch to switch or switch to network connections Cisco s E amp M interface is an RJ 48 connector that allows connections to PBX trunk lines tie lines FIFO First in first out In data communication FIFO refers to a buffering scheme where the first byte of data entering the buffer is the first byte retrieved by the CPU In telephony FIFO refers to a queuing scheme where the first calls received are the first calls processed FXO Foreign Exchange Office An FXO interface connects to the PSTN s central office and is the interface offered on a standard telephone Cisco s FXO interface is an RJ 11 connector that allows an analog connection to be directed at the PSTN s central office This interface is of value for off premise extension applications FXS Foreign Exchange Station An FXS interface connects directly to a standard telephone and supplies ring voltage and dial tone Cisco s FXS interface is an RJ 11 connector that allows connections to basic telephone service equipment keysets and PBXs Multilink PPP Multilink Point to Point Protocol This protocol is a method of splitting recombining and sequencing datagrams across multiple logical data links VC 14 Voice Video and H
11. configure a VoIP peer The number value of the dial peer voice voip command is a tag that uniquely identifies the dial peer To configure the identified VoIP peer use the following commands in dial peer configuration mode Step Command Purpose 1 destination pattern string Define the destination telephone number associated with this VoIP dial peer 2 session target ipv4 destination address Specify a destination IP address for this dns host name dial peer For additional VoIP dial peer configuration options refer to the Voice Related Commands section of the Voice Video and Home Applications Command Reference For examples of how to configure dial peers refer to the section Voice over IP Configuration Examples Configuring Voice over IP for the Cisco 3600 Series VC 31 Optimize Dial Peer and Network Interface Configurations Validation Tips You can check the validity of your dial peer configuration by performing the following tasks Troubleshooting Tips If you have relatively few dial peers configured you can use the show dial peer voice command to verify that the data configured is correct Use this command to display a specific dial peer or to display all configured dial peers Use the show dialplan number command to show the dial peer to which a particular number destination pattern resolves If you are having trouble connecting a call and you suspect the problem is associated with dial peer
12. kbps line Apply adaptive traffic shaping to both DLCIs Use RSVP or IP Precedence to prioritize voice traffic Use compressed RTP to minimize voice packet size Use weighted fair queuing to manage voice traffic Lower MTU size Voice packets are generally small By lowering the MTU size for example to 300 bytes large data packets can be broken up into smaller data packets that can more easily be interwoven with voice packets Note Some applications do not support a smaller MTU size If you decide to lower MTU size use the ip mtu command this command affects only IP traffic Voice Video and Home Applications Configuration Guide Frame Relay for Voice over IP Configuration Example Note Lowering the MTU size affects data throughput speed CIR equal to line rate Make sure that the data rate does not exceed the CIR This is accomplished through generic traffic shaping Use IP Precedence to prioritize voice traffic Use compressed RTP to minimize voice packet header size Traffic shaping Use adaptive traffic shaping to throttle back the output rate based on the BECN If the feedback from the switch is ignored packets both data and voice might be discarded Because the Frame Relay switch does not distinguish between voice and data packets voice packets could be discarded which would result in a deterioration of voice quality Use compressed RTP reduced MTU size and adap
13. over IP this is an IP network VoIP peers point to specific VoIP devices Four call legs make comprise and end to end call two from the perspective of the source router as shown in Figure 6 and two from the perspective of the destination router as shown in Figure 7 A dial peer is associated with each one of these call legs Dial peers are used to apply attributes to call legs and to identify call origin and destination Attributes applied to a call leg include QoS CODEC VAD and fax rate Figure 6 Dial Peer Call Legs from the Perspective of the Source Router Source Destination gt IP cloud Source router yo a j Call leg for POTS Call leg for VolP a dial peer 1 dial peer 2 S Figure 7 Dial Peer Call Legs from the Perspective of the Destination Router Call leg for VoIP Call leg for POTS dial peer 3 dial peer 4 RT T AN a eal i nm Destination router 10354 lt Destination Source Inbound versus Outbound Dial Peers Dial peers are used for both inbound and outbound call legs It is important to remember that these terms are defined from the router s perspective An inbound call leg originates outside the router An outbound call leg originates from the router For inbound call legs a dial peer might be associated to the calling number or the port designation Outbound call legs always have a dial peer associated with them The destination pattern is used to identify the outbound dial peer The
14. source and destination as illustrated in Figure 8 enter the following commands on router 10 1 2 2 dial peer voice 1 pots destination pattern 1408555 port 1 0 0 dial peer voice 2 voip destination pattern 1310555 session target ipv4 10 1 1 2 In the previous configuration example the last four digits in the VoIP dial peer s destination pattern were replaced with wildcards This means that from access server 10 1 2 2 calling any number string that begins with the digits 1310555 will result in a connection to access server 10 1 1 2 This implies that access server 10 1 1 2 services all numbers beginning with those digits From access server 10 1 1 2 calling any number string that begins with the digits 1408555 will result in a connection to access server 10 1 2 2 This implies that access server 10 1 2 2 services all numbers beginning with those digits For more information about stripping and adding digits see the Outbound Dialing on POTS Peers section Figure 9 shows how to complete the end to end call between dial peer 1 and dial peer 4 Configuring Voice over IP for the Cisco 3600 Series VC 27 Configure Dial Peers Figure 9 Outgoing Calls from the Perspective of POTS Dial Peer 2 Destination Source lt IP cloud Dial peer 1 Vor Dial peer 2 Dial peer 3 Dial peer 4 oice port Voice port Ley 1 0 0 Yap 01 22 0112 fy 1 0 0 Len E a 408 555 4000 810 555 1000 POTS call leg VoIP cal
15. to associate a specific logical interface with a POTS peers In addition you can specify direct inward dialing for a POTS peer by using the direct inward dial command Voice Video and Home Applications Configuration Guide Configure IP Networks for Real Time Voice Traffic b VoIP Dial peer describing the characteristics of a packet network connection in the case of Voice over IP this is an IP network VoIP peers point to specific VoIP devices To minimally configure a VoIP peer you need to configure the following two characteristics associated destination telephone number and a destination IP address Use the destination pattern command to define the destination telephone number associated with a VoIP peer Use the session target command to specify a destination IP address for a VoIP peer Refer to the Configure Dial Peers section additional information about configuring dial peers and dial peer characteristics 5 Optimize Dial Peer and Network Interface Configurations You can use VoIP peers to define characteristics such as IP precedence additional QoS parameters when RSVP is configured CODEC and VAD Use the ip precedence command to define IP precedence If you have configured RSVP use either the req qos or acc qos command to configure QoS parameters Use the codec command to configure specific voice coder rates Use the vad command to disable voice activation detection and the transmission of silence packets Refer t
16. voip session target ipv4 172 19 10 10 destination pattern 13085551000 To configure virtual trunk connections in Voice over IP use the connection trunk command The following conditions must be met for Voice over IP to support virtual trunk connections Use the following voice port combinations E amp M to E amp M same type FXS to FXO FXS to FXS with no signaling Do not perform number expansion on the destination pattern telephone numbers configured for trunk connection Configure both end routers for trunk connections The connected Cisco routers must be Cisco 2600 or Cisco 3600 series routers The Cisco AS5300 does not currently support trunk connections Note Because virtual trunk connections do not support number expansion the destination patterns on each side of the trunk connection must match exactly VoIP establishes the trunk connection immediately after it is configured Both ports on either end of the connection are dedicated until you disable trunking for that connection If for some reason the link between the two switching systems goes down the virtual trunk re establishes itself after the link comes back up VC 36 Voice Video and Home Applications Configuration Guide Configure Voice over IP for Microsoft NetMeeting Configure a Trunk Connection To configure virtual trunk connections in a VoIP network use the following commands beginning in global configuration mode Step Comm
17. C 43 Voice over IP Configuration Examples voice port 1 0 1 signal immediate operation 4 wire type 2 Configure the serial interface interface serial 0 0 description serial interface type dce provides clock clock rate 2000000 ip address 172 16 1 123 no shutdown Configuration for Router SLC hostname saltlake Configure pots dial peer 1 dial peer voice 1 pots destination pattern 119 port 1 0 0 Configure pots dial peer 2 dial peer voice 2 pots destination pattern 119 port 1 0 1 Configure voip dial peer 3 dial peer voice 3 voip destination pattern 555 session target ipv4 172 16 1 123 Configure the E amp M interface voice port 1 0 0 signal immediate operation 4 wire type 2 voice port 1 0 0 signal immediate operation 4 wire type 2 Configure the serial interface interface serial 0 0 description serial interface type dte ip address 172 16 65 182 no shutdown Note PBXs should be configured to pass all DTMF signals to the router We recommend that you do not configure store and forward tone Note If you change the gain or the telephony port make sure that the telephony port still accepts DTMF signals VC 44 Voice Video and Home Applications Configuration Guide PSTN Gateway Access Using FXO Connection PSTN Gateway Access Using FXO Connection The following example shows how to configure Voice over IP to link users with the PSTN gateway using an FXO conn
18. Jose offices are using E amp M Port Type II with four wire operation and ImmediateStart signaling Each E amp M interface connects to the router using two voice interface connections Users in San Jose dial 8 569 and then the extension number to reach a destination in Salt Lake City Users in Salt Lake City dial 4 527 and then the extension number to reach a destination in San Jose Figure 13 illustrates the topology of this connection example Figure 13 Linking PBX Users with E amp M Trunk Lines Example 172 16 1 123 172 16 65 182 Dial peer Voice port Voice port Dial peer 1 POTS 1 0 0 1 0 0 1 POTS Le E IP cloud PBX Router SJ Router SLC PBX _ foes Dial peer Voice port Voice port Dial peer Viz 2 POTS 1 0 1 1 0 1 2 POTS San Jose Salt Lake City 408 801 S6616 Note This example assumes that the company already has established a working IP connection between its two remote offices Configuration for Router SJ hostname sanjose Configure pots dial peer 1 dial peer voice 1 pots destination pattern 555 port 1 0 0 Configure pots dial peer 2 dial peer voice 2 pots destination pattern 555 port 1 0 1 Configure voip dial peer 3 dial peer voice 3 voip destination pattern 119 session target ipv4 172 16 65 182 Configure the E amp M interface voice port 1 0 0 signal immediate operation 4 wire type 2 Configuring Voice over IP for the Cisco 3600 Series V
19. OTS call legs As shown in Figure 10 incoming means from the perspective of the router In this case it is the call leg coming into the access server to be forwarded through to the appropriate destination pattern Configuring Voice over IP for the Cisco 3600 Series VC 29 Configure Dial Peers Figure 10 Incoming and Outgoing POTS Call Legs Cisco 3600 Cisco 3600 PBX PBX p ys 27 eS a call leg aie 15564 Unless otherwise configured when a call arrives on the access server the server presents a dial tone to the caller and collects digits until it can identify the destination dial peer After the dial peer has been identified the call is forwarded through the next call leg to the destination There are cases where it might be necessary for the server to use the called number DNIS to find a dial peer for the outgoing call leg for example if the switch connecting the call to the server has already collected the digits DID enables the server to match the called number with a dial peer and then directly place the outbound call With DID the server does not present a dial tone to the caller and does not collect digits it forwards the call directly to the configured destination To use DID and incoming called number a dial peer must be associated with the incoming call leg Before doing this it helps if you understand the logic behind the algorithm used to associate the incoming call leg with the dial peer The
20. a voice call it selects an outbound dial peer by comparing the called number the full E 164 telephone number in the call information with the number configured as the destination pattern for the POTS peer The router then strips out the left justified numbers corresponding to the destination pattern matching the called number If you have configured a prefix the prefix will be put in front of the remaining numbers creating a dial string which the router will then dial If all numbers in the destination pattern are stripped out the user will receive depending on the attached equipment a dial tone For example suppose there is a voice call whose E 164 called number is 1 310 555 2222 If you configure a destination pattern of 1310555 and a prefix of 9 the router will strip out 1310555 from the E 164 telephone number leaving the extension number of 2222 It will then append the prefix 9 to the front of the remaining numbers so that the actual numbers dialed is 9 2222 The comma in this example means that the router will pause for one second between dialing the 9 and the 2 to allow for a secondary dial tone For additional POTS dial peer configuration options refer to the Voice Related Commands section of the Voice Video and Home Applications Command Reference Direct Inward Dial for POTS Peers Direct inward dial DID is used to determine how the called number is treated for incoming P
21. algorithm used to associate incoming call legs with dial peers uses three inputs which are derived from signaling and interface information associated with the call and four defined dial peer elements The three signaling inputs are Called number DNIS Set of numbers representing the destination which is derived from the ISDN setup message or CAS DNIS Calling number ANI Set of numbers representing the origin which is derived from the ISDN setup message or CAS DNIS Voice port The voice port carrying the call The four defined dial peer elements are Destination pattern A pattern representing the phone numbers to which the peer can connect Answer address A pattern representing the phone numbers from which the peer can connect Incoming called number A pattern representing the phone numbers that associate an incoming call leg to a peer based on the called number or DNIS Port The port through which calls to this peer are placed Using the elements the algorithm is as follows For all peers where call type VoIP versus POTS match dial peer type if the type is matched associate the called number with the incoming called number else if the type is matched associate calling number with answer address else if the type is matched associate calling number with destination pattern else if the type is matched associate voice port to port This algorithm shows that if a value is not configured for
22. and Purpose 1 dial peer voice number pots Enter dial peer configuration mode and define a tag number for a POTS dial peer 2 destination pattern string Specify the telephone number associated with the POTS dial peer 3 port slot number subunit number port Associate the POTS dial peer with a specific voice port on the Cisco end router 4 dial peer voice number voip Define a tag number for a VoIP dial peer 5 session target ipv4 destination address Identify the IP address of the appropriate port on the destination end router 6 destination pattern string Identify the destination pattern telephone number of the VoIP dial peer call leg on the destination end router 7 exit Exit dial peer configuration mode 8 configure terminal Enter global configuration mode 9 voice port Enter voice port configuration mode slot number sub unit number port 10 connection trunk string Specify a straight tie line connection virtual trunk connection The string argument refers to the destination pattern telephone number configured for the destination VoIP dial peer The value you configure for the connection trunk command must exactly match the value configured for the VoIP dial peer Note This configuration must be performed on both end routers for the trunk connection to be established Configure Voice over IP for Microsoft NetMeeting Voice over IP can be used with Microsoft NetMeeting Version 2
23. answer address the origin address is used because in most cases the origin address and answer address are the same VC 30 Voice Video and Home Applications Configuration Guide Configure VoIP Peers To configure DID for a particular POTS dial peer use the following commands beginning in global configuration mode Step Command Purpose 1 dial peer voice number pots Enter the dial peer configuration mode to configure a POTS peer 2 direct inward dial Specify direct inward dial for this POTS peer Note Direct inward dial is configured for the calling POTS dial peer For additional POTS dial peer configuration options refer to the Voice Related Commands section of the Voice Video and Home Applications Command Reference Configure VolP Peers Once again VoIP peers enable outgoing calls to be made from a particular telephony device To configure a VoIP peer you need to uniquely identify the peer by assigning it a unique tag number define its destination telephone number and destination IP address As with POTS peers under most circumstances the default values for the remaining dial peer configuration commands will be adequate to establish connections To enter the dial peer configuration mode and select VoIP as the method of voice related encapsulation use the following command in global configuration mode Command Purpose dial peer voice number voip Enter the dial peer configuration mode to
24. call is associated with the outbound dial peer at setup time VC 26 Voice Video and Home Applications Configuration Guide Inbound versus Outbound Dial Peers POTS peers associate a telephone number with a particular voice port so that incoming calls for that telephone number can be received and outgoing calls can be placed VoIP peers point to specific devices by associating destination telephone numbers with a specific IP address so that incoming calls can be received and outgoing calls can be placed Both POTS and VoIP peers are needed to establish Voice over IP connections Establishing communication using Voice over IP is similar to configuring an IP static route you are establishing a specific voice connection between two defined endpoints As shown in Figure 8 for outgoing calls from the perspective of the POTS dial peer 1 the POTS dial peer establishes the source via the originating telephone number or voice port of the call The VoIP dial peer establishes the destination by associating the destination phone number with a specific IP address Figure 8 Outgoing Calls from the Perspective of POTS Dial Peer 1 Source Destination gt IP cloud Dial peer 1 Dial peer 2 Dial peer 3 Dial peer 4 Voice port p 1 emp Voice port 1 0 0 f lt _ 10 1 2 2 10112 4 0 0 iy a 4 i pete fee 310 555 1000 408 555 4000 BOTS call leg VoIP call leg 6613 To configure call connectivity between the
25. ce ports for example in band DTMF digits after the call setup is complete is also trapped by the session application at either end of the connection and carried over the IP network encapsulated in RTCP using the RTCP APP extension mechanism 8 When either end of the call hangs up the RSVP reservations are torn down if RSVP is used and the session ends Each end becomes idle waiting for the next off hook condition to trigger another call setup List of Terms ACOM Term used in G 165 General Characteristics of International Telephone Connections and International Telephone Circuits Echo Cancellers ACOM is the combined loss achieved by the echo canceller which is the sum of the Echo Return Loss Echo Return Loss Enhancement and nonlinear processing loss for the call Call leg A logical connection between the router and either a telephony endpoint over a bearer channel or another endpoint using a session protocol CIR Committed information Rate The average rate of information transfer a subscriber for example the network administrator has stipulated for a Frame Relay PVC CODEC coder decoder Device that typically uses pulse code modulation to transform analog signals into a digital bit stream and digital signals back into analog signals In Voice over IP it specifies the voice coder rate of speech for a dial peer Dial peer An addressable call endpoint In Voice over IP there are two kinds of dial peers POTS and
26. ection In this example users connected to Router SJ in San Jose California can reach PSTN users in Salt Lake City Utah via Router SLC Router SLC in Salt Lake City is connected directly to the PSTN through an FXO interface Figure 14 illustrates the topology of this connection example Figure 14 IP cloud Router SJ i 1 408 555 4000 172 16 1 123 Voice port San Jose 1 0 0 PSTN Gateway Access Using FXO Connection Example PSTN user Router SLC Em 172 16 65 182 Voice port 1 0 0 Salt Lake City S6617 Note This example assumes that the company already has established a working IP connection between its two remote offices Configuration for Router SJ Configure pots dial peer 1 dial peer voice 1 pots destination pattern 14085554000 port 1 0 0 Configure voip dial peer 2 dial peer voice 2 voip destination pattern 9 session target ipv4 172 16 65 182 Configure the serial interface interface serial 0 0 clock rate 2000000 ip address 172 16 1 123 no shutdown Configuring Voice over IP for the Cisco 3600 Series VC 45 Voice over IP Configuration Examples Configuration for Router SLC Configure pots dial peer 1 dial peer voice 1 pots destination pattern 9 port 1 0 0 Configure voip dial peer 2 dial peer voice 2 voip destination pattern 14085554000 session target ipv4 172 16 1 123 Configure serial interface interface serial 0 0 ip address 172
27. elay requirements of real time voice traffic small real time packets which are not multilink encapsulated are transmitted between fragments of the large packets The interleaving feature also provides a special transmit queue for the smaller delay sensitive packets enabling them to be transmitted earlier than other flows Interleaving provides the delay bounds for delay sensitive voice packets on a slow link that is used for other best effort traffic Note Interleaving applies only to interfaces that can configure a multilink bundle interface These include virtual templates dialer interfaces and Integrated Services Digital Network ISDN Basic Rate Interface BRI or Primary Rate Interface PRD interfaces In general Multilink PPP with interleaving is used in conjunction with weighted fair queuing and RSVP or IP Precedence to ensure voice packet delivery Use Multilink PPP with interleaving and weighted fair queuing to define how data will be managed use RSVP or IP Precedence to give priority to voice packets You should configure Multilink PPP if the following conditions exist in your network Point to point connection using PPP Encapsulation Slow links Note Multilink PPP should not be used on links greater than 2 Mbps Multilink PPP support for interleaving can be configured on virtual templates dialer interfaces and ISDN BRI or PRI interfaces To configure interleaving you need to complete the following tasks
28. est Effort Router 2 11 1408116 VoIP IPV4 10 1 1 1 G 729 Best Effort 4 Disa 1729555 POTS Voice Video and Home Applications Configuration Guide Configure POTS Peers Configure POTS Peers Once again POTS peers enable incoming calls to be received by a particular telephony device To configure a POTS peer you need to uniquely identify the peer by assigning it a unique tag number define its telephone number s and associate it with a voice port through which calls will be established Under most circumstances the default values for the remaining dial peer configuration commands will be sufficient to establish connections To enter the dial peer configuration mode and select POTS as the method of voice related encapsulation use the following command in global configuration mode Command Purpose dial peer voice number pots Enter the dial peer configuration mode to configure a POTS peer The number value of the dial peer voice pots command is a tag that uniquely identifies the dial peer This number has local significance only To configure the identified POTS peer use the following commands in dial peer configuration mode Step Command Purpose 1 destination pattern string Define the telephone number associated with this POTS dial peer 2 port slot numberlsubunit numberlport Associate this POTS dial peer with a specific voice port Outbound Dialing on POTS Peers When a router receives
29. etMeeting application Microsoft NetMeeting will open the call dialog box From the Call dialog box select call using H 323 gateway Enter the telephone number in the Address field Click Call to initiate a call to the Cisco 3600 series router from Microsoft NetMeeting Voice over IP Configuration Examples The actual Voice over IP configuration procedure you complete depends on the actual topology of your voice network The following configuration examples should give you a starting point Of course these configuration examples would need to be customized to reflect your network topology Configuration procedures are supplied for the following scenarios FXS to FXS Connection Using RSVP Linking PBX Users with E amp M Trunk Lines PSTN Gateway Access Using FXO Connection PSTN Gateway Access Using FXO Connection PLAR Mode These examples are described in the following sections FXS to FXS Connection Using RSVP The following example shows how to configure Voice over IP for simple FXS to FXS connections In this example a very small company consisting of two offices has decided to integrate Voice over IP into its existing IP network One basic telephony device is connected to Router RLB 1 therefore Router RLB 1 has been configured for one POTS peer and one VoIP peer Router RLB w and Router VC 38 Voice Video and Home Applications Configuration Guide FXS to FXS Connection Using RSVP R12 e establish the WAN con
30. face Encapsulation method is Frame Relay Configuring Voice over IP for the Cisco 3600 Series VC 23 Configure Number Expansion Fair queuing is enabled IP RTP header compression is enabled The subinterface has been configured as follows MTU size is inherited from the main interface IP address for the subinterface is specified Bandwidth is set to 64 kbps Generic traffic shaping is enabled with 32 kbps CIR where Bc 4000 bits and Be 4000 bits Frame Relay DLCI number is specified IP RTP header compression is enabled Note When traffic bursts over the CIR output rate is held at the speed configured for the CIR for example traffic will not go beyond 32 kbps if CIR is set to 32 kbps For more information about Frame Relay refer to the Configuring Frame Relay chapter in the Wide Area Networking Configuration Guide Configure Number Expansion In most corporate environments the telephone network is configured so that you can reach a destination by dialing only a portion an extension number of the full E 164 telephone number Voice over IP can be configured to recognize extension numbers and expand them into their full E 164 dialed number by using two commands in tandem destination pattern and num exp Before you configure these two commands it is helpful to map individual telephone extensions with their full E 164 dialed numbers This task can be done easily by creating a number e
31. for Real Time Voice Traffic You need to have a well engineered network end to end when running delay sensitive applications such as VoIP Fine tuning your network to adequately support VoIP involves a series of protocols and features geared toward Quality of Service QoS It is beyond the scope of this document to explain the specific details relating to wide scale QoS deployment Cisco IOS software provides many tools for enabling QoS on your backbone such as Random Early Detection RED Weighted Random Early Detection WRED Fancy queuing meaning custom priority or weighted fair queuing and IP Precedence To configure your IP network for real time voice traffic you need to take into consideration the entire scope of your network then select the appropriate QoS tool or tools Configuring Voice over IP for the Cisco 3600 Series VC 17 Configure IP Networks for Real Time Voice Traffic The important thing to remember is that QoS must be configured throughout your network not just on the Cisco 3600 series devices running VoIP to improve voice network performance Not all QoS techniques are appropriate for all network routers Edge routers and backbone routers in your network do not necessarily perform the same operations the QoS tasks they perform might differ as well To configure your IP network for real time voice traffic you need to take into consideration the functions of both edge and backbone routers in your network then select
32. higher bandwidth requirements for voice For example to specify a CODEC rate of G 711a law for VoIP dial peer 108 enter the following dial peer voice 108 voip destination pattern 14085551234 codec g71llalaw session target ipv4 10 0 0 8 Voice Video and Home Applications Configuration Guide Configure Voice over IP using a Trunk Connection Configure VAD for a VoIP Dial Peer To disable the transmission of silence packets for a selected VoIP peer use the following commands beginning in global configuration mode Step Command Purpose 1 dial peer voice number voip Enter the dial peer configuration mode to configure a VoIP peer 2 vad Disable the transmission of silence packets enabling VAD The default for the vad command is enabled normally the default configuration for this command is the most desirable If you are operating on a high bandwidth network and voice quality is of the highest importance you should disable vad Using this value will result in better voice quality but it will also require higher bandwidth requirements for voice For example to enable VAD for VoIP dial peer 108 enter the following dial peer voice 108 voip destination pattern 14085551234 vad session target ipv4 10 0 0 8 Configure Voice over IP using a Trunk Connection A trunk is a communication line between two switching systems typically the switching equipment in a central office and a PBX A trunk connection is a pe
33. igured on both Router 1 and Router 2 Configure Number Expansion To define how to expand an extension number into a particular destination pattern use the following command in global configuration mode Command Purpose num exp extension number extension string Configure number expansion You can verify the number expansion information by using the show num exp command to verify that you have mapped the telephone numbers correctly After you have configured dial peers and assigned destination patterns to them you can verify number expansion information by using the show dialplan number command to see how a telephone number maps to a dial peer Configuring Voice over IP for the Cisco 3600 Series VC 25 Configure Dial Peers Configure Dial Peers The key point to understanding how Voice over IP functions is to understand dial peers Each dial peer defines the characteristics associated with a call leg as shown in Figure 6 and Figure 7 A call leg is a discrete segment of a call connection that lies between two points in the connection All the call legs for a particular connection have the same connection ID There are two different kinds of dial peers POTS Dial peer describing the characteristics of a traditional telephony network connection POTS peers point to a particular voice port on a voice network device VoIP Dial peer describing the characteristics of a packet network connection in the case of Voice
34. ions number Specify the total number of RTP header compression connections supported on an interface RTP Header Compression Configuration Example The following example enables RTP header compression for a serial interface interface 0 ip rtp header compression encapsulation ppp ip rtp compression connections 25 For more information about RTP header compression see the Configuring IP Multicast Routing chapter of the Network Protocols Configuration Guide Part 1 Configuring Voice over IP for the Cisco 3600 Series VC 21 Configure Frame Relay for Voice over IP Configure Custom Queuing Some QoS features such as IP RTP reserve and custom queuing are based on the transport protocol and the associated port number Real time voice traffic is carried on UDP ports ranging from 16384 to 16624 This number is derived from the following formula 16384 4 number of voice ports in the Cisco 3600 series router Custom Queuing and other methods for identifying high priority streams should be configured for these port ranges For more information about custom queuing refer to the Performing Basic System Management chapter in the Configuration Fundamentals Configuration Guide Configure Weighted Fair Queuing Weighted fair queuing ensures that queues do not starve for bandwidth and that traffic gets predictable service Low volume traffic streams receive preferential service high volume traffic streams share the remaining ca
35. l leg To complete the end to end call between dial peer 1 and dial peer 4 as illustrated in Figure 9 enter the following commands on router 10 1 1 2 dial peer voice 4 pots destination pattern 1310555 port 1 0 0 dial peer voice 3 voip destination pattern 1408555 session target ipv4 10 1 2 2 Create a Peer Configuration Table There is specific data relative to each dial peer that needs to be identified before you can configure dial peers in Voice over IP One way to do this is to create a peer configuration table VC 28 Using the example in Figure 5 Router 1 with an IP address of 10 1 1 1 connects a small sales branch office to the main office through Router 2 There are three telephones in the sales branch office that need to be established as dial peers Router 2 with an IP address of 10 1 1 2 is the primary gateway to the main office as such it needs to be connected to the company s PBX There are four devices that need to be established as dial peers in the main office all of which are basic telephones connected to the PBX Figure 5 shows a diagram of this small voice network Table 6 shows the peer configuration table for the example illustrated in Figure 5 S6614 Table 6 Peer Configuration Table for Sample Voice Over IP Network Commands Dial Peer Tag Ext Dest Pattern Type Voice Port session target CODEC QoS Router 1 1 6 1408116 POTS 10 1729555 VoIP IPV4 10 1 1 2 G 729 B
36. lan and decided how to integrate it into your existing IP network you are ready to configure your network devices to support Voice over IP Configuring Voice over IP for the Cisco 3600 Series VC 15 Voice over IP Configuration Task List Voice over IP Configuration Task List VC 16 To configure Voice over IP on the Cisco 3600 series you need to complete the following tasks 1 Configure IP Networks for Real Time Voice Traffic Configure your IP network to support real time voice traffic Fine tuning your network to adequately support VoIP involves a series of protocols and features geared toward Quality of Service QoS To configure your IP network for real time voice traffic you need to take into consideration the entire scope of your network then select and configure the appropriate QoS tool or tools a Multilink PPP with Interleaving b RTP Header Compression c Custom Queuing d Weighted Fair Queuing Refer to Configure IP Networks for Real Time Voice Traffic section for information about how to select and configure the appropriate QoS tools to optimize voice traffic on your network 2 Configure Frame Relay for Voice over IP Optional If you plan to run Voice over IP over Frame Relay you need to take certain factors into consideration when configuring Voice over IP for it to run smoothly over Frame Relay For example a public Frame Relay cloud provides no guarantees for QoS Refer to the Configure Frame
37. ndset this signals an off hook condition to the signaling application part of Voice over IP in the Cisco 3600 series router 2 The session application part of Voice over IP issues a dial tone and waits for the user to dial a telephone number 3 The user dials the telephone number those numbers are accumulated and stored by the session application 4 After enough digits are accumulated to match a configured destination pattern the telephone number is mapped to an IP host via the dial plan mapper The IP host has a direct connection to either the destination telephone number or a PBX that is responsible for completing the call to the configured destination pattern Configuring Voice over IP for the Cisco 3600 Series VC 13 List of Terms 5 The session application then runs the H 323 session protocol to establish a transmission and a reception channel for each direction over the IP network If the call is being handled by a PBX the PBX forwards the call to the destination telephone If RSVP has been configured the RSVP reservations are put into effect to achieve the desired quality of service over the IP network 6 The CODECs are enabled for both ends of the connection and the conversation proceeds using RTP UDP IP as the protocol stack 7 Any call progress indications or other signals that can be carried in band are cut through the voice path as soon as end to end audio channel is established Signaling that can be detected by the voi
38. nection between the two offices Because one POTS telephony device is connected to Router RLB 2 it has also been configured for only one POTS peer and one VoIP peer Note In this example only the calling end Router RLB 1 is request RSVP Figure 12 illustrates the topology of this FXS to FXS connection example Figure 12 FXS to FXS Connection Example Serial port IP cloud Serial port 1 0 1 3 1 3 1 0 Ce 64 kbps Router RLB w 128 kbps 64 kbps Voice port Voice port 1 0 0 1 0 0 LY Dial peer 1 Dial peer 2 POTS POTS Serial port Serial port 0 0 1 0 S6612 Configuration for Router RLB 1 hostname rlb 1 Create voip dial peer 10 dial peer voice 10 voip Define its associated telephone number and IP address destination pattern 4155554000 session target ipv4 40 0 0 1 Request RSVP req qos guaranteed delay Create pots dial peer 1 dial peer voice 1 pots Define its associated telephone number and voice port destination pattern 4085554000 port 1 0 0 Configure serial interface 0 0 interface Serial0 0 ip address 10 0 0 1 255 0 0 0 no ip mroute cache Configure RTP header compression ip rtp header compression ip rtp compression connections 25 Enable RSVP on this interface ip rsvp bandwidth 48 48 fair queue 64 256 36 Configuring Voice over IP for the Cisco 3600 Series VC 39 Voice over IP Configuration Examples clockrate 64000 router igrp network 10 net
39. o the Optimize Dial Peer and Network Interface Configurations section for additional information about optimizing dial peer characteristics 6 Configure Voice Ports You need to configure your Cisco 3600 series router to support voice ports In general voice port commands define the characteristics associated with a particular voice port signaling type voice ports on the Cisco 3600 series support three basic voice signaling types a FXO Foreign Exchange Office interface b FXS The Foreign Exchange Station interface c E amp M The Ear and Mouth interface or RecEive and TransMit interface Under most circumstances the default voice port command values are adequate to configure FXO and FXS ports to transport voice data over your existing IP network Because of the inherent complexities involved with PBX networks E amp M ports might need specific voice port values configured depending on the specifications of the devices in your telephony network For information about configuring voice ports refer to the Configuring Voice Ports chapter 7 Configure Voice over IP for Microsoft NetMeeting Optional Voice over IP can be used with Microsoft NetMeeting Version 2 x when the Cisco 3600 series router is used as the voice gateway Refer to the Configure Voice over IP for Microsoft NetMeeting section for more information about configuring Voice over IP to support Microsoft NetMeeting Configure IP Networks
40. ome Applications Configuration Guide Prerequisite Tasks PBX Private Branch Exchange Privately owned central switching office PLAR Private Line Auto Ringdown This type of service results in a call attempt to some particular remote endpoint when the local extension is taken off key POTS Plain Old Telephone Service Basic telephone service supplying standard single line telephones telephone lines and access to the public switched telephone network POTS dial peer Dial peer connected via a traditional telephony network POTS peers point to a particular voice port on a voice network device PSTN Public Switched Telephone Network PSTN refers to the local telephone company PVC Permanent virtual circuit QoS Quality of Service QoS refers to the measure of service quality provided to the user RSVP Resource Reservation Protocol This protocol supports the reservation of resources across an IP network Trunk Service that allows quasi transparent connections between two PBXs a PBX and a local extension or some other combination of telephony interfaces to be permanently conferenced together by the session application and signaling passed transparently through the IP network VoIP dial peer Dial peer connected via a packet network in the case of Voice over IP this is an IP network VoIP peers point to specific VoIP devices Prerequisite Tasks Before you can configure your Cisco 3600 series router to use Voice o
41. pacity obtaining equal or proportional bandwidth In general weighted fair queuing is used in conjunction with Multilink PPP with interleaving and RSVP or IP Precedence to ensure that voice packet delivery Use weighted fair queuing with Multilink PPP to define how data will be managed use RSVP or IP Precedence to give priority to voice packets For more information about weighted fair queuing refer to the Performing Basic System Management chapter in the Configuration Fundamentals Configuration Guide Configure Frame Relay for Voice over IP VC 22 You need to take certain factors into consideration when configuring Voice over IP for it to run smoothly over Frame Relay A public Frame Relay cloud provides no guarantees for QoS For real time traffic to be transmitted in a timely manner the data rate must not exceed the committed information rate CIR or there is the possibility that packets will be dropped In addition Frame Relay traffic shaping and RSVP are mutually exclusive This is particularly important to remember if multiple DLCIs are carried on a single interface For Frame Relay links with slow output rates less than or equal to 64 kbps where data and voice are being transmitted over the same PVC we recommend the following solutions Separate DLCIs for voice and data By providing a separate subinterface for voice and data you can use the appropriate QoS tool per line For example each DLCI would use 32 kbps of a 64
42. peer 2 acc qos best effort controlled load Specify the QoS value below which an SNMP guaranteed delay trap will be generated Note RSVP reservations are only one way If you configure RSVP the VoIP dial peers on both ends of the connection must be configured for RSVP Configure CODEC and VAD for Dial Peers Coder decoder CODEC and voice activity detection VAD for a dial peer determine how much bandwidth the voice session uses CODEC typically is used to transform analog signals into a digital bit stream and digital signals back into analog signals in this case it specifies the voice coder rate of speech for a dial peer VAD is used to disable the transmission of silence packets Configure CODEC for a VoIP Dial Peer VC 34 To specify a voice coder rate for a selected VoIP peer use the following commands beginning in global configuration mode Step Command Purpose 1 dial peer voice number voip Enter the dial peer configuration mode to configure a VoIP peer 2 codec g711alaw g711ulaw g729r8 Specify the desired voice coder rate of speech The default for the codec command is g729r8 normally the default configuration for this command is the most desirable If however you are operating on a high bandwidth network and voice quality is of the highest importance you should configure the codec command for g711alaw or ulaw Using this value will result in better voice quality but it will also require
43. recedence value Configure RSVP for Dial Peers If you have configured your WAN or LAN interfaces for RSVP you must configure the QoS for any associated VoIP peers To configure quality of service for a selected VoIP peer use the following commands starting in global configuration mode Step Command Purpose 1 dial peer voice number voip Enter the dial peer configuration mode to configure a VolP peer 2 req qos best effort Specify the desired quality of service to be used controlled load guaranteed delay Note We suggest that you select controlled load for the requested quality of service For example to specify guaranteed delay QoS for VoIP dial peer 108 enter the following dial peer voice 108 voip destination pattern 14085551234 req qos controlled load session target ipv4 10 0 0 8 In this example every time a connection is made through VoIP dial peer 108 an RSVP reservation request is made between the local router all intermediate routers in the path and the final destination router Configuring Voice over IP for the Cisco 3600 Series VC 33 Optimize Dial Peer and Network Interface Configurations To generate an SNMP trap message if the reserved QoS is less than the configured value for a selected VoIP peer use the following commands beginning in global configuration mode Step Command Purpose 1 dial peer voice number voip Enter the dial peer configuration mode to configure a VoIP
44. rmanent physical layer wire point to point connection Voice over IP simulates a trunk connection by creating virtual trunk tie lines between PBXs connected to Cisco 2600 and 3600 series routers on each side of a VoIP connection See Figure 11 In this example two PBXs are connected using a virtual trunk PBX A is connected to Router A via an E amp M voice port PBX B is connected to Router B via an E amp M voice port The Cisco routers spoof the connected PBXs into believing that a permanent trunk tie line exists between them Figure 11 Virtual Trunk Connection 1 308 555 0180 172 19 10 10 172 20 10 10 ae 7 Virtual trunk connection Configuring Voice over IP for the Cisco 3600 Series VC 35 Configure Voice over IP using a Trunk Connection The routers on both sides of the Voice over IP connection must be configured for trunk connections For the scenario described in Figure 11 configure Router A to support trunk connections as follows configure terminal voice port 1 0 0 connection trunk 15105554000 dial peer voice 10 pots destination pattern 13085551000 port 1 0 0 dial peer voice 100 voip session target ipv4 172 20 10 10 destination pattern 15105554000 For the scenario described in Figure 11 configure Router B to support trunk connections as follows configure terminal voice port 1 0 0 connection trunk 13085551000 dial peer voice 20 pots destination pattern 15105554000 port 1 0 0 dial peer voice 200
45. the appropriate QoS tool or tools In general edge routers perform the following QoS functions Packet classification Admission control Bandwidth management Queuing In general backbone routers perform the following QoS functions High speed switching and transport Congestion management Queue management Scalable QoS solutions require cooperative edge and backbone functions Note In a subsequent Cisco IOS release we have implemented enhancements to improve QoS on low speed wide area links such as ISDN MLPPP and Frame Relay running on edge routers For more information about these enhancements refer to the Cisco IOS Release 12 0 5 T IP RTP feature module Although not mandatory some QoS tools have been identified as being valuable in fine tuning your network to support real time voice traffic To configure your IP network for QoS using these tools perform one or more of the following tasks Configure Multilink PPP with Interleaving Configure RTP Header Compression Configure Custom Queuing Configure Weighted Fair Queuing Each of these components is discussed in the following sections VC 18 Voice Video and Home Applications Configuration Guide Configure Multilink PPP with Interleaving Configure Multilink PPP with Interleaving Multi class Multilink PPP Interleaving allows large packets to be multilink encapsulated and fragmented into smaller packets to satisfy the d
46. tive traffic shaping based on BECN to hold data rate to CIR Use generic traffic shaping to obtain a low interpacket wait time For example set Bc to 4000 to obtain an inter packet wait of 125 ms Note We recommend FRF 12 fragmentation setup rules for Voice over IP connections over Frame Relay FRF 12 was implemented in the Cisco IOS Release 12 0 4 T For more information refer to the Cisco IOS Release 12 0 4 T Voice over Frame Relay using FRF 11 and FRF 12 feature module Frame Relay for Voice over IP Configuration Example For Frame Relay it is customary to configure a main interface and several subinterfaces one subinterface per PVC The following example configures a Frame Relay main interface and a subinterface so that voice and data traffic can be successfully transported interface Serial0 0 ip mtu 300 no ip address encapsulation frame relay no ip route cache no ip mroute cache fair queue 64 256 1000 frame relay ip rtp header compression interface Serial0 0 1 point to point ip mtu 300 ip address 40 0 0 7 255 0 0 0 no ip route cache no ip mroute cache bandwidth 64 traffic shape rate 32000 4000 4000 frame relay interface dlci 16 frame relay ip rtp header compression In this configuration example the main interface has been configured as follows MTU size of IP packets is 300 bytes No IP address is associated with this serial interface The IP address must be assigned for the subinter
47. ver IP you must first Establish a working IP network For more information about configuring IP refer to the IP Overview Configuring IP Addressing and Configuring IP Services chapters in the Network Protocols Configuration Guide Part 1 Install the one slot or two slot NM 1V NM 2V voice network module into the appropriate bay of your Cisco router For more information about the physical characteristics of the voice network module or how to install it refer to the installation documentation Voice Network Module and Voice Interface Card Configuration Note that came with your voice network module Complete your company s dial plan Establish a working telephony network based on your company s dial plan Integrate your dial plan and telephony network into your existing IP network topology Merging your IP and telephony networks depends on your particular IP and telephony network topology In general we recommend the following suggestions Use canonical numbers wherever possible It is important to avoid situations where numbering systems are significantly different on different routers or access servers in your network Make routing and or dialing transparent to the user for example avoid secondary dial tones from secondary switches where possible Contact your PBX vendor for instructions about how to reconfigure the appropriate PBX interfaces After you have analyzed your dial p
48. ving and a maximum real time traffic delay of 20 milliseconds and then applies that virtual template to the Multilink PPP bundle interface virtual template 1 ppp multilink encapsulated ppp ppp multilink interleave ppp multilink fragment delay 20 ip rtp reserve 16384 100 64 multilink virtual template 1 Configure RTP Header Compression Real Time Transport Protocol RTP is used for carrying packetized audio traffic over an IP network RTP header compression compresses the IP UDP RTP header in an RTP data packet from 40 bytes to approximately 2 to 4 bytes most of the time as shown in Figure 4 This compression feature is beneficial if you are running Voice over IP over slow links Enabling compression on both ends of a low bandwidth serial link can greatly reduce the network overhead if there is a lot of RTP traffic on that slow link Typically an RTP packet has a payload of approximately 20 to 160 bytes for audio applications that use compressed payloads RTP header compression is especially beneficial when the RTP payload size is small for example compressed audio payloads between 20 and 50 bytes Figure 4 RTP Header Compression Before RTP header compression 20 bytes 8 bytes 12 bytes IP UDP RTP Payload SSS Header lt 20 to 160 bytes gt After RTP header compression 2 to 4 bytes xe IP UDP RTP header 20 to 160 bytes gt Payload 12076 VC 20 Voice Video and
49. work 20 network 40 oOo ooo oOo o0OoOo o O oO O VC 40 Voice Video and Home Applications Configuration Guide FXS to FXS Connection Using RSVP Configuration for Router RLB w hostname rlb w Configure serial interface 1 0 interface Seriall1 0 ip address 10 0 0 2 255 0 0 0 Configure RTP header compression ip rtp header compression ip rtp compression connections 25 Enable RSVP on this interface ip rsvp bandwidth 96 96 fair queue 64 256 3 Configure serial interface 1 3 interface Seriall 3 ip address 20 0 0 1 255 0 0 0 Configure RTP header compression ip rtp header compression ip rtp compression connections 25 Enable RSVP on this interface ip rsvp bandwidth 96 96 fair queue 64 256 3 Configure IGRP router igrp 888 network 10 0 0 network 20 0 0 0 0 network 40 0 0 0 Configuration for Router R12 e hostname r12 e Configure serial interface 1 0 interface Seriall1 0 ip address 40 0 0 2 25 0 0 0 Configure RTP header compression ip rtp header compression ip rtp compression connections 25 Enable RSVP on this interface ip rsvp bandwidth 96 96 fair queue 64 256 3 Configure serial interface 1 3 interface Seriall 3 ip address 20 0 0 2 255 0 0 0 Configure RTP header compression ip rtp header compression ip rtp compression connections 25 Enable RSVP on this interface ip rsvp bandwidth 96 96 fair queue 64 256 3 clockrate 128000 Configuring Voice over IP for the Cisco
50. x when the Cisco 3600 or Cisco 2600 series router is used as the voice gateway Use the latest version of DirectX drivers from Microsoft on your PC to improve the voice quality of NetMeeting Configure Voice over IP to Support Microsoft NetMeeting To configure Voice over IP to support NetMeeting create a VoIP peer that contains the following information Session Target IP address or DNS name of the PC running NetMeeting e CODEC g71lulaw or g71lalaw Configuring Voice over IP for the Cisco 3600 Series VC 37 Voice over IP Configuration Examples Configure Microsoft NetMeeting for Voice over IP To configure NetMeeting to work with Voice over IP complete the following steps Step 1 Step 2 Step 3 Step 4 Step 5 Step 6 Step 7 Step 8 Step 9 From the Tools menu in the NetMeeting application select Options NetMeeting will display the Options dialog box Click the Audio tab Click the Calling a telephone using NetMeeting check box Enter the IP address of the Cisco AS5300 in the IP address field Under General click Advanced Click the Manually configured compression settings check box Select the CODEC value CCITT ulaw 8000Hz Click the Up button until this CODEC value is at the top of the list Click OK to exit Initiate a Call Using Microsoft NetMeeting To initiate a call using Microsoft NetMeeting perform the following steps Step 1 Step 2 Step 3 Step 4 Click the Call icon from the N
51. xpansion table Create a Number Expansion Table In Figure 5 a small company wants to use Voice over IP to integrate its telephony network with its existing IP network The destination pattern or expanded telephone number associated with Router 1 located to the left of the IP cloud are 408 115 xxxx 408 116 xxxx and 408 117 xxxx where xxxx identifies the individual dial peers by extension The destination pattern or expanded telephone number associated with Router 2 located to the right of the IP cloud is 729 555 xxxx VC 24 Voice Video and Home Applications Configuration Guide Configure Number Expansion Figure 5 Sample Voice over IP Network 729 555 1001 729 555 1002 408 115 1001 BSL Uy 729 555 1000 729 555 1003 T1 Cisco 3600 ISDN PRI 408 116 1002 Voice port Router 1 l h a O D 7 WAN Voice port 0 D Lep D WAN Ga EE 5 10 1 1 2 c T1 ISDN PRI Cisco 3600 Router 2 a g wo 408 117 1003 Table 5 shows the number expansion table for this scenario Table 5 Sample Number Expansion Table Extension Destination Pattern Num Exp Command Entry Syal 40811 num exp 5 408115 O 40811 num exp 6 408116 Tress 40811 num exp 7 408117 Ty 729555 num exp 2 729555 Note You can use the period symbol to represent variables such as extension numbers in a telephone number The information included in this example needs to be conf

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