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C O N T E N T S About This Guide xi Objectives xi Audience xi Cisco IOS Software Documentation xi Organization xiv Command Syntax Conventions xiv Cisco Connection Online xv Documentation Feedback xv CHA PTER 1 Voice over IP Overview 1-1 Voice Primer 1-1 How VoIP Processes a Typical Telephone Call 1-2 Numbering Scheme 1-2 Analog Compared with Digital 1-3 CODECs 1-3 Mean Opinion Score 1-3 Delay 1-4 Jitter 1-5 End-to-End Delay 1-5 Echo 1-5 Signaling 1-6 Cisco 1751 Router Software Configuration Guide OL-10
Contents CHA PTER 2 VoIP Configuration 2-1 Prerequisite Tasks 2-1 Configuration Tasks 2-2 Configure IP Networks for Real-Time Voice Traffic 2-2 Configure RSVP for Voice 2-3 Enable RSVP 2-3 RSVP Configuration Example 2-4 Configure Multilink PPP with Interleaving 2-4 Multilink PPP Configuration Example 2-5 Configure RTP Header Compression 2-6 Enable RTP Header Compression on a Serial Interface 2-7 Change the Number of Header Compression Connections 2-7 RTP Header Compression Configuration Example 2-7 Confi
Contents Configure FXS or FXO Voice Ports 2-15 Verifying Your Configuration 2-16 Troubleshooting Tips 2-16 Fine-Tune FXS and FXO Voice Ports 2-16 Configure E&M Voice Ports 2-18 Verifying Your Configuration 2-19 Troubleshooting Tips 2-20 Fine-Tune E&M Voice Ports 2-20 Additional VoIP Dial Peer Configurations 2-21 Configure IP Precedence for Dial Peers 2-22 Configure RSVP for Dial Peers 2-22 Configure CODEC and VAD for Dial Peers 2-23 Configure CODEC for a VoIP Dial Peer 2-23 Configure VAD for a VoIP Dial Pe
Contents Linking PBX Users with E&M Trunk Lines 3-5 Router SJ Configuration 3-6 Router SLC Configuration 3-7 FXO Gateway to PSTN 3-7 Router SJ Configuration 3-8 Router SLC Configuration 3-8 FXO Gateway to PSTN (PLAR Mode) 3-9 Router SJ Configuration 3-9 Router SLC Configuration 3-10 CHA PTER 4 VoIP Commands 4-1 acc-qos 4-4 answer-address 4-5 codec 4-6 comfort-noise 4-7 connection 4-8 cptone 4-10 description 4-11 destination-pattern 4-12 dial-control-mib 4-13 dial-peer voice 4-13 dial-type 4-14 echo-canc
Contents impedance 4-20 input gain 4-21 ip precedence 4-22 ip udp checksum 4-22 music-threshold 4-23 non-linear 4-24 num-exp 4-25 operation 4-25 output attenuation 4-26 port 4-27 prefix 4-28 req-qos 4-29 ring frequency 4-30 ring number 4-31 session protocol 4-32 session target 4-32 show call active voice 4-34 show call history voice 4-37 show controllers voice 4-40 show diag 4-42 show dial-peer voice 4-45 show dialplan incall number 4-47 show dialplan number 4-48 show num-exp 4-48 show voice dsp 4-49 show
Contents shutdown (voice-port configuration) 4-56 signal 4-56 snmp enable peer-trap poor-qov 4-58 snmp-server enable traps 4-59 snmp trap link-status 4-60 timeouts initial 4-61 timeouts interdigit 4-62 timing 4-63 type 4-65 vad 4-67 voice-port 4-67 CHA PTER 5 VoIP Debug Commands 5-1 Using Debug Commands 5-1 debug voip ccapi error 5-2 debug voip ccapi inout 5-2 debug vpm all 5-5 debug vpm dsp 5-5 debug vpm error 5-6 debug vpm port 5-6 debug vpm signal 5-7 debug vpm spi 5-8 debug vtsp all 5-10 debug vtsp
Contents debug vtsp stats 5-19 debug vtsp tone 5-20 debug vtsp vofr subframe 5-20 CHA PTER 6 Routing Between Virtual LANs Overview 6-1 What Is a VLAN? 6-1 LAN Segmentation 6-2 Security 6-2 Broadcast Control 6-3 Performance 6-3 Network Management 6-3 Communication Between VLANs 6-3 VLAN Colors 6-3 Why Implement VLANs? 6-4 Communicating Between VLANs 6-4 VLAN Translation 6-4 Designing Switched VLANs 6-4 CHA PTER 7 Configuring Routing Between VLANs with IEEE 802.1Q Encapsulation 7-1 IEEE 802.
Contents Assigning IP Address to Network Interface 7-3 Configuring IPX Routing over IEEE 802.1Q 7-4 Enabling NetWare Routing 7-4 Defining the VLAN Encapsulation Format 7-4 Configuring NetWare on the Subinterface 7-4 IEEE 802.1Q Encapsulation Configuration Examples 7-5 Configuring AppleTalk over IEEE 802.1Q Example 7-5 Configuring IP Routing over IEEE 802.1Q Example 7-5 Configuring IPX Routing over IEEE 802.
About This Guide This section discusses the objectives, audience, conventions, and organization of the Cisco 1751 Router Software Configuration Guide and provides general information about Cisco IOS software documentation. Cisco documentation and additional literature are available in a CD-ROM package that ships with your product. The Documentation CD-ROM, a member of the Cisco Connection Family, is updated monthly. Therefore, it might be more up to date than printed documentation.
Two master indexes provide indexing information for the Cisco IOS software documentation set: an index for the configuration guides and an index for the command references. In addition, individual books contain a book-specific index.
Figure 1 Cisco IOS Software Documentation Modules Module FC Configuration Guide Module FR Command Reference Module FC/FR: Configuration Fundamentals • Access Server and Router Product Overview • Cisco IOS Software Configuration Basics • Images and Configuration Files • Interface Configuration • System Management Module P1C Configuration Guide Module P1R Command Reference Module P1C/P1R: Network Protocols, Part 1 • IP Addressing • IP Services • IP Routing Protocols Module P3C Configuration Guide Mod
Organization Table 1 describes the contents of each chapter in this document. Table 1 Organization Chapter Title Description Chapter 1 Voice over IP Overview Overview of the VoIP software application and, for those unfamiliar with telephony, a brief Voice Primer.
Table 2 Command Syntax Guide Convention Description boldface screen font Examples of information that you must enter. < > Nonprinting characters, such as passwords, appear in angled brackets. [ ] Default responses to system prompts appear in square brackets. Cisco Connection Online Cisco Connection Online (CCO) is Cisco Systems’ primary, real-time support channel. Maintenance customers and partners can self-register on CCO to obtain additional information and services.
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1 Voice over IP Overview Voice over IP (VoIP) enables a Cisco 1751 router (hereafter referred to as the router) to carry voice traffic (for example, telephone calls and faxes) over an IP network. Cisco’s voice support is implemented using voice packet technology. In VoIP, the digital signal processor (DSP) segments the voice signal into frames and stores them in voice packets.
How VoIP Processes a Typical Telephone Call Before configuring VoIP on your router, it helps to understand what happens at an application level when you place a call using VoIP. The general flow of a two-party voice call using VoIP is as follows: 1. The user picks up the handset; this signals an off-hook condition to the signaling application part of VoIP in the router. 2. The session application part of VoIP issues a dial tone and waits for the user to dial a telephone number. 3.
Analog Compared with Digital Analog transmission is not particularly robust or efficient at recovering from line noise. Because analog signals degrade over distance, they need to be periodically amplified; this amplification boosts both the voice signal and ambient line noise, resulting in degradation of the quality of the transmitted sound.
Table 1 Compression Methods and MOS Scores Compression Method Bit Rate (kbps) MOS Score G.711 PCM 64 4.1 G.723.1 MP-MLQ 6.3 3.9 G.723.1 ACELP 5.3 3.65 G.726 ADPCM 32 3.85 G.729 CS-ACELP 1 8 3.92 G.729 x 2 Encodings 8 3.27 G.729 x 3 Encodings 8 2.68 G.729a CS-ACELP 8 3.7 1.
Table 2 CODEC-Induced Delays CODEC Bit Rate (kbps) Compression Delay (ms) G.729 CS-ACELP 8 15 G.729a CS-ACELP 8 15 Another handling delay is the time it takes to generate a voice packet. In VoIP, the DSP generates a frame every 10 milliseconds. Two of these frames are then placed within one voice packet; the packet delay is therefore 20 milliseconds. Another source of handling delay is the time it takes to move the packet to the output queue.
In Cisco’s voice implementations, echo cancellers are enabled using the echo-cancel enable command. The echo trails are configured using the echo-cancel-coverage command. VoIP has configurable echo trails of 8, 16, 24, and 32 milliseconds. Signaling Although there are various types of signaling used in telecommunications today, this document describes only those with direct applicability to Cisco’s voice implementations.
2 VoIP Configuration This chapter explains how to configure VoIP on your router and contains the following sections: • Prerequisite Tasks • Configuration Tasks • Configure IP Networks for Real-Time Voice Traffic • Configure Number Expansion • Configure Dial Peers • Configure Voice Ports • Additional VoIP Dial Peer Configurations • Configure Frame Relay for VoIP • Configure Microsoft NetMeeting for VoIP Prerequisite Tasks Before you can configure your router to use VoIP, you need to perform
– Make routing and 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 plan and decided how to integrate it into your existing IP network, you are ready to configure your network devices to support VoIP.
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. Although not mandatory, some QoS tools can be valuable in fine-tuning your network to support real-time voice traffic.
Router(config-if)# ip rsvp bandwidth [interface-kbps] [single-flow-kbps] This command starts RSVP and sets the bandwidth and single-flow limits. The default maximum bandwidth is up to 75 percent of the bandwidth available on the interface. By default, the amount reservable by a flow can be up to the entire reservable bandwidth. On subinterfaces, RSVP applies to the more restrictive of the available bandwidths of the physical interface and the subinterface.
• Note Links slower than 2 Mbps Do not use multilink PPP 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: • Configure the dialer interface or virtual template, as defined in the relevant chapters of the Dial Solutions Configuration Guide for Cisco IOS Release 12.1T.
Configure RTP Header Compression Real-Time Transport Protocol (RTP) is used for carrying audio traffic in packets 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 1. This compression feature is beneficial if you are running VoIP over slow links.
Enable RTP Header Compression on a Serial Interface You need to enable compression on both ends of a serial connection. To enable RTP header compression, use the following interface configuration command: Router(config-if)# ip rtp header-compression [passive] 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.
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. VoIP 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 helps to map individual telephone extensions with their full E.
Table 1 Note Sample Number Expansion Table Extension Destination Pattern Num-Exp Command Entry 1... 14085551... num-exp 1... 14085551... To expand a four-digit extension beginning with the numeral 1 by prefixing 1408555 to it 2... 17295552... num-exp 2... 17295552... To expand a four-digit extension beginning with the numeral 2 by prefixing 1408555 to it 3... 17295553... num-exp 3... 17295553...
Figure 3 Dial Peer Call Legs from the Perspective of the Source Router Source Destination Call leg for POTS dial peer 1 Figure 4 18944 IP cloud Call leg for VoIP dial peer 2 Dial Peer Call Legs from the Perspective of the Destination Router Call leg for VoIP dial peer 3 Call leg for POTS dial peer 4 Destination Source 24418 IP cloud There are basically two different kinds of dial peers with each voice implementation: • POTS—(also known as “plain old telephone service” or “basic telephone ser
POTS dial peer 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 dial 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 dial peers are needed to establish VoIP connections.
To complete the end-to-end call between dial peer 1 and dial peer 4 as illustrated in Figure 6, enter the following commands on router 10.1.1.2: Router(config)# dial-peer voice 4 pots Router(config-dial-peer)# destination-pattern 13105551000 Router(config-dial-peer)# port 0/0 Router(config)# dial-peer voice 3 voip Router(config-dial-peer)# destination-pattern 14085554000 Router(config-dial-peer)# session target ipv4:10.1.2.
Outbound Dialing on POTS Dial Peers When a router receives 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 dial peer. The router then removes the left-justified numbers corresponding to the destination pattern that matches the called number.
Verifying Your Configuration You can check the validity of your dial peer configuration by performing the following tasks: • 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 which dial peer is reached when a particular number is dialed.
The voice ports support three basic voice signaling types: • FXS—The foreign exchange station interface uses a standard RJ-11 modular telephone cable to connect directly to a standard telephone, fax machine, PBXs, or similar device, and supplies ring, voltage, and dial tone to the station. • FXO—The foreign exchange office interface uses a RJ-11 modular telephone cable to connect local calls to a PSTN central office or to PBX that does not support E&M signaling.
Command Required or Optional Step 9 music-threshold number Optional Specify the threshold (in dB) for on-hold music. Valid entries are from –70 to –30 decibels (dB). Step 10 description string Optional Attach descriptive text about this voice-port connection. Step 11 comfort-noise Optional If voice activity detection (VAD) is activated, specify that background noise is generated.
Valid Entries Default Values –6 to 14 dB 0 dB Command Task Step 1 configure terminal Enter the global configuration mode. Step 2 voice-port slot-number/port Identify the voice port you want to configure, and enter the voice port configuration mode. Step 3 input gain value Specify (in dB) the amount of gain to be inserted at the receiver side of the interface. Step 4 output attenuation value Specify (in dB) the amount of 0 to 14 dB attenuation at the transmit side of the interface.
Command Task Valid Entries Default Values Step 12 timing pulse-digit milliseconds (FXO ports only) If the voice-port dial type is pulse, configure the pulse digit signal duration. 10 to 20 ms 20 ms Step 13 timing pulse-inter-digit milliseconds (FXO ports only) If the 100 to 1000 ms 500 ms voice-port dial type is pulse, configure the pulse inter-digit signal duration.
Step 7 Command Required / Optional Task type {1 | 2 | 3 | 5} Required Select the appropriate E&M interface type.
Troubleshooting Tips If you are having trouble connecting a call and you suspect the problem is associated with the voice-port configuration, you can try to resolve the problem by performing the following tasks: • Ping the associated IP address to confirm connectivity. If you cannot ping your destination, refer to the Network Protocols Configuration Guide, Part 1 for Cisco IOS Release 12.1T. • Use the show voice-port command to make sure that the port is enabled.
Task Step 7 non-linear Enable nonlinear processing, which shuts off any signal if no near-end speech is detected. (Nonlinear processing is used with echo-cancellation.) Step 8 timeouts initial seconds Specify the number of 0 to 120 sec seconds the system will wait for the caller to input the first digit of the dialed digits.
Configure IP Precedence for Dial Peers Use the ip precedence command to give voice packets a higher priority than other IP data traffic. The ip precedence command should also be used if RSVP is not enabled and you would like to give voice packets a priority over other IP data traffic. IP precedence scales better than RSVP, but provides no admission control.
To generate a Simple Network Management Protocol (SNMP), use the following commands beginning in global configuration mode: Command Task Step 1 dial-peer voice number voip Enter the dial peer configuration mode to configure a VoIP dial peer. Step 2 acc-qos [best-effort | controlled-load | guaranteed-delay] Generate an SNMP event if the QoS for a dial peer drops below a specified level. Note RSVP reservations are one-way only.
Configure VAD for a VoIP Dial Peer To disable the transmission of silence packets and enable VAD for a selected VoIP dial peer, use the following global configuration commands: Command Task Step 1 dial-peer voice number voip Enter the dial peer configuration mode to configure a VoIP dial peer. Step 2 vad Disable the transmission of silence packets . The default for the vad command is enabled; normally, the default configuration for this command is the most desirable.
– Use RSVP or IP precedence to prioritize voice traffic. – Use compressed RTP to minimize voice packet header size. • Traffic shaping—Use adaptive traffic shaping to slow the output rate based on the backward explicit congestion notification (BECN). If the feedback from the switch is ignored, packets (both data and voice) might be discarded.
Note • MTU size is inherited from the main interface. • IP address for the subinterface is specified. • RSVP is enabled to use the default value, which is 75 percent of the configured bandwidth. • Bandwidth is set to 64 kbps. • Generic traffic shaping is enabled with 32-kbps CIR where committed burst (Bc) = 4000 bits and excess burst (Be) = 4000 bits. • Frame Relay DLCI number is specified. • IP RTP header compression is enabled.
Step 8 Click the Up button until this CODEC value is at the top of the list. Step 9 Click OK to exit. Initiate a Call Using Microsoft NetMeeting To initiate a call using Microsoft NetMeeting, perform the following steps: Step 1 Click the Call icon from the NetMeeting application. Microsoft NetMeeting opens the call dialog box. Step 2 From the Call dialog box, select call using H.323 gateway. Step 3 Enter the telephone number in the Address field.
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3 VoIP Configuration Examples This chapter demonstrates how to configure VoIP in four different scenarios. The actual VoIP configuration procedure depends on the actual topology of your voice network. The following configuration examples should give you a starting point. These configuration examples would need to be customized to reflect your network topology.
Figure 1 FXS-to-FXS Connection Example Serial port 0 1 Serial port 0 1 IP cloud Voice port 0/0 Router RLB-w Serial port 0 128K Router RLB-e 64K Voice port 0/0 Serial port 0 Router RLB-1 Dial peer 1 POTS (408) 555-4001 Router RLB-2 17418 64K Dial peer 2 POTS (415) 555-3001 Configuration for Router RLB-1 hostname RLB-1 ! Create voip dial-peer 2 dial-peer voice 2 voip ! Define its associated telephone number and IP address destination-pattern 14155553001 sess-target ipv4:40.0.0.
Configuration for Router RLB-w hostname RLB-w ! Configure serial interface 0 interface serial0/0 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 interface serial1/0 ip address 20.0.0.1 255.0.0.
Configuration for Router RLB-e hostname RLB-e ! Configure serial interface 0 interface serial0/0 ip address 40.0.0.2 255.0.0.0 ! 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 interface serial1/0 ip address 20.0.0.2 255.0.0.
Configuration for Router RLB-2 hostname RLB-2 ! Create pots dial-peer 2 dial-peer voice 2 pots ! Define its associated telephone number and voice-port destination-pattern 14155553001 port 0/0 ! Create voip dial-peer 1 dial-peer voice 1 voip !Define its associated telephone number and IP address destination-pattern 14085554001 sess-target ipv4:10.0.0.1 ! Configure serial interface 0 interface serial1/0 ip address 40.0.0.1 255.0.0.
Figure 2 Linking PBX Users with E&M Trunk Lines Example 172.16.1.123 Dial peer 1 POTS (408) 555-4001 PBX 172.16.65.182 Voice port 0/0 Voice port 0/0 Router SJ Router SLC Dial peer 3 POTS PBX (801) 555-3001 Dial peer 2 (408) 555-4002 POTS Voice port 0/1 San Jose (408) Note 17419 IP cloud Dial peer 4 POTS (801) 555-3002 Salt Lake City (801) Voice port 0/1 This example assumes that the company has already established a working IP connection between its two remote offices.
Router SLC Configuration hostname router SLC !Configure pots dial-peer 3 dial-peer voice 3 pots destination-pattern 1801555.... port 0/0 !Configure pots dial-peer 4 dial-peer voice 4 pots destination-pattern 1801555.... port 0/1 !Configure voip dial-peer 1 dial-peer voice 1 voip destination-pattern 1408555.... session target ipv4:172.16.1.
Figure 3 FXO Gateway to PSTN Example PSTN user Router SLC Router SJ PSTN cloud IP cloud 1(408) 555-4000 1(801) . . . . . . . San Jose Note Voice port 0/0 172.16.1.123 Voice port 0/0 Salt Lake City 18943 172.16.65.182 This example assumes that the company has already established a working IP connection between its two remote offices.
FXO Gateway to PSTN (PLAR Mode) The following example shows an FXO gateway to PSTN connection in 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 4 illustrates the topology of this connection example.
Router SLC Configuration hostname router SLC ! Configure pots dial-peer 1 dial-peer voice 1 pots destination-pattern 1801....... port 0/0 ! Configure voip dial-peer 2 dial-peer voice 2 voip destination-pattern 14085554000 session target ipv4:172.16.1.123 ip precedence 5 ! Configure the voice port voice port 0/0 connection plar 14085554000 ! Configure the serial interface 0 interface serial1/0 ip address 172.16.65.182 255.255.0.
4 VoIP Commands This chapter provides an alphabetical listing of all of the VoIP commands that are new or specific to the Cisco 1751 router. All other commands used with this feature are documented in the Cisco IOS Release 12.1T command reference documents. Table 1 lists and describes the commands in this chapter that are used to configure and monitor VoIP. Table 1 Commands Used to Configure and Monitor VoIP Command Description acc-qos Generate an SNMP event if the QoS drops below a specified level.
Table 1 Commands Used to Configure and Monitor VoIP Command Description ip udp checksum Calculate the UDP checksum for voice packets transmitted by the dial peer. music-threshold Specify the threshold for on-hold music for a specified voice port. non-linear Enable nonlinear processing in the echo canceller. num-exp Define how to expand an extension number into a particular destination pattern. operation Select a specific cabling scheme for E&M ports.
Table 1 Commands Used to Configure and Monitor VoIP Command Description vad Enable VAD for the calls using this dial peer. voice-port Enter the voice port configuration mode. A subset of the commands listed are voice-port commands. Different voice signaling types support different voice-port commands. Table 2 lists the router voice-port commands and the signaling types supported.
Table 2 Router Voice-Port Commands and Signaling Types Supported (Continued) Voice-Port Command FXO FXS E&M pulse X – X pulse-inter-digit X – X wink-duration – – X wink-wait – – X – – X type acc-qos To generate an SNMP event if the QoS for a dial peer drops below a specified level, use the acc-qos dial-peer configuration command. Use the no form of this command to use the default value for this feature.
Example The following example selects guaranteed-delay as the specified level below which an SNMP trap message is generated: dial-peer voice 10 voip acc-qos guaranteed-delay Related Commands req-qos answer-address To specify the full E.164 telephone number to be used to identify the dial peer of an incoming call, use the answer-address dial-peer configuration command. Use the no form of this command to disable this feature.
For calls coming in from a POTS interface, the answer-address command is not used to select an incoming dial peer. The incoming POTS dial peer is selected on the basis of the port configured for that dial peer. This command applies to both VoIP and POTS dial peers. Note The Cisco IOS software does not check the validity of the E.164 telephone number; it accepts any series of digits as a valid number. Example The following example configures the E.
Syntax Description g711alaw G.711 A-Law 64,000 bits per second (bps). g711ulaw G.711 U-Law 64,000 bps. g723ar53 G.723.1 ANNEX-A 5,300 bps. g723ar63 G.723.1 ANNEX-A 6,300 bps. g723r53 G.723.1 5,300 bps. g723r63 G.723.1 6,300 bps. g726r16 G.726 16,000 bps. g726r24 G.726 24,000 bps. g726r32 G.726 32,000 bps. g729br8 G.729 ANNEX-B 8,000 bps. g729r8 G.729 8,000 bps. Default g729r8. Command Mode Dial-peer configuration.
Example The following example configures a voice coder rate that provides toll quality and uses a relatively high amount of bandwidth: dial-peer voice 10 voip codec g711alaw comfort-noise To specify whether or not background noise should be generated, use the comfort-noise voice-port configuration command. Use the no form of this command to disable this feature. comfort-noise no comfort-noise Syntax Description This command has no arguments or keywords. Default Enabled.
connection {plar | trunk } string no connection {plar | trunk } string Syntax Description plar Private line auto ringdown (PLAR) connection. PLAR connection associates a dial peer directly with an interface; when an interface goes off-hook, the dial peer sets up the second call leg and creates a conference call without the caller having to dial any digits. trunk Straight tie-line connection to a private branch exchange (PBX). string Destination telephone number.
cptone To configure a voice call progress tone locale, use the cptone voice-port configuration command. Use the no form of this command to disable this feature. cptone {australia | brazil | china | finland | france | germany | japan | northamerica | unitedkingdom} no cptone Syntax Description australia Analog voice interface-related default tone, ring, and cadence setting for Australia. brazil Analog voice interface-related default tone, ring, and cadence setting for Brazil.
Usage Guidelines Use the cptone command to specify a regional analog voice interface-related tone, ring, and cadence setting for a specified voice port. This command only affects the tones generated at the local interface. It does not affect any information passed to the remote end of a connection or any tones generated at the remote end of a connection.
destination-pattern To specify either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer, use the destination-pattern dial-peer configuration command. Use the no form of this command to disable this feature. destination-pattern [+]string no destination-pattern Syntax Description string Series of digits that specify the E.
dial-control-mib To specify attributes for the call history table, use the dial-control-mib global configuration command. dial-control-mib {max-size number | retain-timer number} Syntax Description max-size number Maximum size of the call history table. Valid entries are from 0 to 500 table entries. A value of 0 prevents any history from being retained. retain-timer number Length of time, in minutes, for entries in the call history table. Valid entries are from 0 to 2147483647 minutes.
Syntax Description number Digit(s) defining a particular dial peer. Valid entries are from 1 to 2147483647. voip VoIP dial peer using voice encapsulation on the POTS network. pots POTS dial peer using VoIP encapsulation on the IP backbone. Default No dial peer configuration mode is preconfigured. Command Mode Global configuration. Usage Guidelines Use the dial-peer voice global configuration command to switch to the dial peer configuration mode from the global configuration mode.
Syntax Description dtmf Touch-tone dialer. pulse Pulse dialer. Default dtmf. Command Mode Voice-port configuration. Usage Guidelines Use the dial-type command to specify an out-dialing type for an FXO or E&M voice-port interface; this command does not apply to FXS voice ports because they do not generate out-dialing. Voice ports can always detect DTMF and pulse signals. This command does not affect voice-port dialing detection. The dial-type command affects out-dialing as configured for the dial peer.
Usage Guidelines Use the echo-cancel coverage command to adjust the coverage size of the echo canceller. This command enables cancellation of voice that is sent out of the interface and received back on the same interface within the configured amount of time. If the local loop (the distance from the analog interface to the connected equipment producing the echo) is longer, the configured value of this command should be extended.
Usage Guidelines The echo-cancel command enables cancellation of voice that is sent out of the interface and is received back on the same interface. Disabling echo cancellation might cause the remote side of a connection to hear an echo. Because echo cancellation is an invasive process that can minimally degrade voice quality, this command should be disabled if it is not needed. The echo-cancel command does not affect the echo heard by the user on the analog side of the connection.
Usage Guidelines VoIP monitors the quality of voice received over the network. Use the expect-factor command to specify when the router generates an SNMP trap to the network manager. This command only applies to VoIP peers. Example The following example configures toll quality of voice when connecting to a dial peer: dial-peer voice 10 voip expect-factor 0 fax-rate To establish the rate at which a fascimile (fax) is sent to the specified dial peer, use the fax-rate dial-peer configuration command.
The values for this command apply only to the fax transmission speed and do not affect the quality of the fax itself. The higher values provide a faster transmission speed but monopolize a significantly larger portion of the available bandwidth. Slower transmission speeds use less bandwidth. If the fax-rate command is set above the codec command rate in the same dial peer, the data sent over the network for fax transmission exceeds the bandwidth reserved for RVSP.
Example The following example disables the icpif command: dial-peer voice 10 voip icpif 0 impedance To specify the terminating impedance of a voice-port interface, use the impedance voice-port configuration command. Use the no form of this command to restore the default value. impedance {600c | 600r | 900c | complex1 | complex2} no impedance Syntax Description 600c 600 ohms complex. 600r 600 ohms real. 900c 900 ohms complex. complex1 Complex 1. complex2 Complex 2. Default 600 ohms.
Example The following example configures an FXO voice port for a terminating impedance of 600 ohms: voice port 0/0 impedance 600r input gain To configure a specific input gain value, use the input gain voice-port configuration command. Use the no form of this command to disable this feature. input gain value no input gain value Syntax Description value Amount of gain in decibels (dB) to be inserted at the receiver side of the interface. Acceptable value is any integer from –6 to 14. Default 0 dB.
ip precedence To set IP precedence (priority) for packets sent by the dial peer, use the ip precedence dial-peer configuration command. Use the no form of this command to restore the default value for this command. ip precedence number no ip precedence Syntax Description number Integer specifying the IP precedence value. Valid entries are 0 to 7. A value of 0 means that no precedence (priority) has been set. Default No precedence (0). Command Mode Dial-peer configuration.
Default Disabled. Command Mode Dial-peer configuration. Usage Guidelines Use the ip udp checksum command to enable UDP checksum calculation for each outbound voice packet. This command is disabled by default to speed up the transmission of the voice packets. If you suspect that the connection has a high error rate, you should enable ip udp checksum to prevent bad voice packets forwarded to the DSP. This command only applies to VoIP peers.
If the value for this command is set too high, VAD interprets music-on-hold as silence, and the remote end does not hear the music. If the value for this command is set too low, VAD compresses and passes silence when the background is noisy, creating unnecessary voice traffic.
num-exp To define how to expand an extension number into a particular destination pattern, use the num-exp global configuration command. num-exp extension-number expanded-number Syntax Description extension-number Digit(s) defining an extension number for a particular dial peer. expanded-number Digit(s) defining the expanded telephone number or destination pattern for the extension number listed. Default No number expansions are predefined. Command Mode Global configuration.
operation {2-wire | 4-wire} no operation {2-wire | 4-wire} Syntax Description 2-wire Two-wire E&M cabling scheme. 4-wire Four-wire E&M cabling scheme. Default 2-wire. Command Mode Voice-port configuration. Usage Guidelines The operation command only affects voice traffic. Signaling is independent of two-wire versus four-wire settings. If the wrong cable scheme is specified, the user might get voice traffic in only one direction.
Syntax Description value Amount of attenuation in dB at the transmit side of the interface. Acceptable value is any integer from 0 to 14. Default 0 dB. Command Mode Voice-port configuration. Usage Guidelines A system-wide loss plan must be implemented by using both input gain and output attenuation commands. Other equipment (including PBXs) in the system must be taken into account when creating a loss plan.
Syntax Description slot-number Slot number in the router where the VIC is installed. Valid entries are from 0 to 2, depending on the slot where it has been installed. port Voice port. Valid entries are 0 or 1. Default No port is preconfigured. Command Mode Dial-peer configuration. Usage Guidelines Use the port configuration command to associate the designated voice port with the selected dial peer.
Command Mode Dial-peer configuration. Usage Guidelines Use the prefix command to specify a prefix for a specific dial peer. When an outgoing call is initiated to this dial peer, the prefix string value is first sent to the telephony interface, before the telephone number is associated with the dial peer. If you want to configure different prefixes for dialed numbers on the same interface, you need to configure different dial peers. This command only applies to POTS peers.
Command Mode Dial-peer configuration. Usage Guidelines Use the req-qos command to request a specific QoS to be used in reaching a dial peer. This command is like acc-qos; the software reserves a certain amount of bandwidth to provide the selected QoS. Cisco IOS software uses RSVP to request QoS guarantees from the network. This command only applies to VoIP peers.
Example The following example configures the ring frequency for 50 Hz: voice port 0/0 ring frequency 50 Related Commands ring number ring number To specify the number of rings for a specified FXO voice port, use the ring number voice-port configuration command. Use the no form of this command to reset the default value for this command. ring number number no ring number number Syntax Description number Number of rings detected before answering the call. Valid entries are numbers from 1 to 10.
Related Commands ring frequency session protocol To establish a session protocol for calls between the local and remote routers via the packet network, use the session protocol dial-peer configuration command. Use the no form of this command to reset the default value for this command. session protocol cisco no session protocol Syntax Description cisco Cisco Session Protocol. Default cisco. Command Mode Dial-peer configuration.
Syntax Description ipv4:destination-address IP address of the dial peer. dns:host-name Domain name system (DNS) server is used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device. (Optional) You can use one of the following wildcards with this keyword when defining the session target for VoIP dial peers: • $s$.—Source destination pattern is used as part of the domain name. • $d$.
The following example configures a session target using dns and the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. means that the router uses the unmatched portion of the dialed number—in this case, the four-digit extension—to identify the dial peer. As in the previous example, the domain is cisco.com. dial-peer voice 10 voip destination-pattern 1310222....
Figure 1 Call Legs Example Call leg for VoIP dial peer 3 Call leg for POTS dial peer 4 Destination Source 24418 IP cloud These two call legs are associated by the connection ID. The connection ID is global across the voice network so that you can associate two call legs on one router with two call legs on another router, thereby providing an end-to-end view of a call.
Table 3 Show-Call-Active-Voice Command Field Descriptions (Continued) Field Description ConnectTime Time at which the call was connected. Dial-Peer Tag of the dial peer transmitting this call. ERLLevel Current Echo Return Loss (ERL) level for this call. FaxTxDuration Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.
Table 3 Show-Call-Active-Voice Command Field Descriptions (Continued) Field Description RemoteIPAddress Remote system IP address for the VoIP call. RemoteUDPPort Remote system UDP listener port to which voice packets are transmitted. RoundTripDelay Voice packet round trip delay between the local and remote system on the IP backbone during the call. SelectedQoS Selected RSVP QoS for the call.
Usage Guidelines Use the show call history voice privileged EXEC command to display the call history table. The call history table contains a listing of all calls connected through this router in descending time order since VoIP was enabled. You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword last, and define the number of calls to be displayed with the argument number.
Table 4 Show-Call-History-Voice Command Field Descriptions (Continued) Field Description GapFillWithPrediction Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call.
Table 4 Show-Call-History-Voice Command Field Descriptions (Continued) Field Description TransmitBytes Number of bytes transmitted by this peer during the call. TransmitPackets Number of packets transmitted by this peer during the call. TxDuration Duration of the transmit path open from this peer to the voice gateway for the call. VADEnable Whether or not VAD was enabled for this call. VoiceTxDuration Duration of voice transmitted from this peer to voice gateway for this call.
Sample Display The following is sample output from the show controllers voice command: router# show controllers voice EPIC Switch registers: STDA 0xFF STDB 0x0 SARA 0x0 SARB 0xFF SAXA 0xFF SAXB 0x0 STCR 0x3F MFAIR 0x3F STAR 0x65 OMDR 0xE2 VNSR 0x0 PMOD 0x4C PBNR 0xFF POFD 0xF0 POFU 0x18 PCSR 0x1 PICM 0x0 CMD1 0xA0 CMD2 0x70 CBNR 0xFF CTAR 0x2 CBSR 0x20 CSCR 0x0 DSP 0 Host Port Interface: HPI Control Register 0x202 InterfaceStatus 0x2A MaxMessageSize 0x80 RxRingBufferSize 0x6 TxRingBufferSize 0x9 pInsertRx 0
Tx Message 4: packet_length 66 channel_id 0 0000:0000 0000 0000 0000 003D 0020:0000 0006 0006 0006 0006 0040:0006 0006 0006 0006 0006 --More-Tx Message 5: packet_length 66 channel_id 0 0000:0000 0000 0000 0000 003E 0020:0000 0006 0006 0006 0006 0040:0006 0006 0006 0006 0006 packet_id 003A 0000 0006 0006 0006 0006 198 process id1 0xFECE process id2 0xFACE 0000 0000 0000 0006 0006 0006 0000 packet_id 003B 0000 0006 0006 0006 0006 198 process id1 0xFECE process id2 0xFACE 0000 0000 0000 0006 0006 0006 0000
Syntax Description This command contains no arguments or keywords. Command Mode Privileged EXEC. Usage Guidelines This command displays information for the electrically erasable programmable read-only memory (EEPROM), motherboard, and the WAN interface cards and voice interface cards (WICs/VICs).
Sample Display The following is sample output from the show diag command: router# show diag Slot 0: C1750 1FE VE Mainboard port adapter, 6 ports Port adapter is analyzed Port adapter insertion time unknown EEPROM contents at hardware discovery: Hardware revision 0.
show dial-peer voice To display configuration information for dial peers, use the show dial-peer voice privileged EXEC command. show dial-peer voice [number] Syntax Description number Displays configuration for the dial peer identified by the argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767. Command Mode Privileged EXEC.
The following is sample output from the show dial-peer voice command for a VoIP dial peer: router# show dial-peer voice 10 VoiceOverIpPeer10 tag = 10, dest-pat = `', incall-number = `14085', group = 0, Admin state is up, Operation state is down Permission is Answer, type = voip, session target = `', sess-proto = cisco, req-qos = bestEffort, acc-qos = bestEffort, fax-rate = voice, codec = g729r8, Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled, Connect Time = 0, Charged Units = 0 Succ
Table 5 Show-Dial-Peer-Voice Command Field Descriptions (Continued) Field Description req-qos Configured requested QoS for calls for this dial peer. session target Session target of this peer. sess-proto Session protocol to be used for Internet calls between local and remote router via the IP backbone. Successful Calls Number of completed calls to this peer. tag Unique dial-peer ID number. VAD Whether or not VAD is enabled for this dial peer.
Example The following example tests whether the telephone extension 57681 can be reached through voice port 0/1: show dialplan incall 0/1 number 57681 Related Commands show dialplan number show dialplan number To show which dial peer is reached when a particular telephone number is dialed, use the show dial plan number privileged EXEC command. show dial plan number dial string Syntax Description dial string Particular destination pattern (telephone number). Command Mode Privileged EXEC.
Syntax Description dialed-number Displays number expansion for the specified dialed number. Command Mode Privileged EXEC. Usage Guidelines Use the show num-exp privileged EXEC command to display all of the number expansions configured for this router. To display number expansion for only one number, specify that number by using the dialed-number argument.
Syntax Description This command has no arguments or keywords. Command Mode Privileged EXEC. Usage Guidelines This command also applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.
Syntax Description slot-number Slot number in the router where the VIC is installed. Valid entries are from 0 to 2, depending on the slot where it has been installed. port Voice port. Valid entries are 0 or 1. Command Mode Privileged EXEC. Usage Guidelines Use the show voice port privileged EXEC command to display configuration and VIC-specific information about a specific port.
Sample Display The following is sample output from the show voice port command for an E&M voice port: router# show voice port 0/0 E&M Slot 0/0 Type of VoicePort is E&M Operation State is unknown Administrative State is unknown The Interface Down Failure Cause is 0 Alias is NULL Noise Regeneration is disabled Non Linear Processing is disabled Music On Hold Threshold is Set to 0 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is disabled Echo Cancel Coverage is set to 16ms Connecti
The following is sample output from the show voice port command for an FXS voice port: router# show voice port 0/0 Foreign Exchange Station 0/0 Slot is 0, Port is 0 Type of VoicePort is FXS Operation State is DORMANT Administrative State is UP The Interface Down Failure Cause is 0 Alias is NULL Noise Regeneration is enabled Non Linear Processing is enabled Music On Hold Threshold is Set to 0 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to
Table 8 Show-Voice-Port Command Field Descriptions (Continued) Field Description Echo Cancellation Whether or not echo cancellation is enabled for this port. Hook Flash Duration Timing Maximum length of hook flash signal. Hook Status Hook status of the FXO/FXS interface. Impedance Configured terminating impedance for the E&M interface. In Gain Amount of gain inserted at the receiver side of the interface. In Seizure Incoming seizure state of the E&M interface.
Table 8 Show-Voice-Port Command Field Descriptions (Continued) Field Description Wink Duration Timing Maximum wink duration for wink start signaling. Wink Wait Duration Timing Maximum wink wait duration for wink start signaling. Related Commands show show show show call active voice call history voice dial-peer voice num-exp shutdown (dial-peer configuration) To change the administrative state of the selected dial peer from up to down, use the shutdown dial-peer configuration command.
shutdown (voice-port configuration) To take the voice ports for a specific VIC offline, use the shutdown voice-port configuration command. Use the no form of this command to put the ports back in service. shutdown no shutdown Syntax Description This command has no arguments or keywords. Default Enabled. Command Mode Voice-port configuration. Usage Guidelines When you enter the shutdown command, all ports on the VIC are disabled, and there is dead silence on the telephone connected to the interface.
Syntax Description loop-start Loop Start signaling. Used for FXO and FXS interfaces. With Loop Start signaling, only one side of a connection can hang up. This is the default setting for FXO and FXS voice ports. ground-start Ground Start signaling. Used for FXO and FXS interfaces. Ground Start allows both sides of a connection to place a call and to hang up.
Example The following example configures ground-start signaling, which means that both sides of a connection can place a call and hang up, as the signaling type for a voice port: configure terminal voice port 1/1 signal ground-start snmp enable peer-trap poor-qov To generate poor-quality-of-voice notification for applicable calls associated with VoIP dial peers, use the snmp enable peer-trap poor-qov dial-peer configuration command. Use the no form of this command to disable this feature.
snmp-server enable traps To enable the router to send SNMP traps, use the snmp-server enable traps global configuration command. Use the no form of this command to disable SNMP traps. snmp-server enable traps [trap-type] [trap-option] no snmp-server enable traps [trap-type] [trap-option] Defaults No traps are enabled. Some trap types cannot be controlled with this command. These traps are either always enabled or enabled by some other means.
The following example enables the router to send all traps to the host myhost.cisco.com using the community string public: snmp-server enable traps snmp-server host myhost.cisco.com public The following example enables the router to send Frame Relay and environmental monitor traps to the host myhost.cisco.com using the community string public: snmp-server enable traps frame-relay snmp-server enable traps envmon temperature snmp-server host myhost.cisco.
If you are managing the equipment with an SNMP manager (such as Maestro), enable this command. Enabling link-status messages allows the SNMP manager to learn of a status change without polling the equipment. If you are not using an SNMP manager, disable this command to avoid unnecessary network traffic.
Example The following example sets the initial digit timeout value to 15 seconds: voice port 0/0 timeouts initial 15 Related Commands timeouts interdigit timing timeouts interdigit To configure the interdigit timeout value for a specified voice port, use the timeouts interdigit voice-port configuration command. Use the no form of this command to restore the default value for this command.
Related Commands timeouts initial timing timing To specify timing parameters (other than those defined by the timeouts commands) for a specified voice port, use the timing voice-port configuration command. Use the no form of this command to reset the default value for this command. timing timing-value no timing timing-value Syntax Description timing-value Table 9 One of the keyword/argument pairs listed in Table 9.
Table 9 Timing Keywords/Arguments, Descriptions, and Valid Entries wink-duration milliseconds The maximum wink signal duration, in milliseconds, for a wink start signal Numbers from 100 to 400 wink-wait milliseconds The maximum wink-wait duration, in milliseconds, for a wink start signal Numbers from 100 to 5000 Default The default values for the timing keywords/arguments are listed in Table 10.
Table 11 Timing Keywords/Arguments Call Signal Directions Timing Keyword/Argument Call Signal Direction dial-pulse min-delay milliseconds In digit milliseconds Out inter-digit milliseconds Out pulse pulses per second Out pulse-inter-digit milliseconds Out wink-duration milliseconds Out wink-wait milliseconds Out Example The following example configures the clear-wait duration to 300 milliseconds: voice port 0/0 timing clear-wait 300 Related Commands timeouts initial timeouts interdigit t
Syntax Description 1 For the following lead configuration: E—Output, relay to ground. M—Input, referenced to ground. 2 For the following lead configuration: E—Output, relay to SG. M—Input, referenced to ground. SB—Feed for M, connected to –48V. SG—Return for E, galvanically isolated from ground. 3 For the following lead configuration: E—Output, relay to ground. M—Input, referenced to ground. SB—Connected to –48V. SG—Connected to ground.
Example The following example selects type 3 as the interface type for your voice port: voice port 0/0 type 3 vad To enable voice activity detection (VAD) for the calls using this dial peer, use the vad dial-peer configuration command. Use the no form of this command to disable this feature. vad no vad Syntax Description This command has no arguments or keywords. Default Enabled. Command Mode Dial-peer configuration. Usage Guidelines Use the vad command to enable VAD.
Syntax Description slot-number Slot number in the router where the VIC is installed. Valid entries are from 0 to 2, depending on the slot where it has been installed. port Voice port. Valid entries are 0 or 1. Default No voice-port mode is configured. Command Mode Global configuration. Usage Guidelines Use the voice-port global configuration command to switch to the voice port configuration mode from the global configuration mode.
5 VoIP Debug Commands This chapter documents debug commands that are new or specific to the Cisco 1751 router. All other commands used with this feature are documented in the Debug Command Reference chapter for the Cisco IOS Release12.1T.
Caution Debugging is assigned a high priority in your router CPU process, and it can render your router unusable. For this reason, use debug commands only to troubleshoot specific problems. The best time to use debug commands is during periods of low network traffic and few users to decrease the likelihood that the debug command processing overhead affects network users.
Usage Guidelines The debug voip ccapi inout EXEC command traces the execution path through the call control API, which serves as the interface between the call session application and the underlying network-specific software. You can use the output from this command to understand how calls are being handled by the router. This command shows how a call flows through the system. Using this debug level, you can see the call setup and teardown operations performed on both the telephony and network call legs.
The following output shows the called party is alerted, a codec is negotiated, and voice path is cut through: cc_api_call_alert(vdbPtr=0x60B6C5D4, callID=0x2, prog_ind=0x8, sig_ind=0x1) sess_appl: ev(6), cid(2), disp(0) ssa: cid(2)st(1)oldst(0)cfid(-1)csize(0)in(0)fDest(0)-cid2(1)st2(1)oldst2(0) ccCallAlert (callID=0x1, prog_ind=0x8, sig_ind=0x1) ccConferenceCreate (confID=0x60B6C150, callID1=0x1, callID2=0x2, tag=0x0) cc_api_bridge_done (confID=0x1, srcIF=0x60B6C5D4, srcCallID=0x2, dstCallID=0x1, dispositi
The following output shows that disconnection is generated from the calling party and that call legs are torn down and disconnected: cc_api_call_disconnected(vdbPtr=0x60BFB530, callID=0x1, cause=0x10) sess_appl: ev(9), cid(1), disp(0) ssa: cid(1)st(5)oldst(3)cfid(1)csize(0)in(1)fDest(0)-cid2(2)st2(5)oldst2(4) ccConferenceDestroy (confID=0x1, tag=0x0) cc_api_bridge_done (confID=0x1, srcIF=0x60B6C5D4, srcCallID=0x2, dstCallID=0x1, disposition=0 tag=0x0) cc_api_bridge_done (confID=0x1, srcIF=0x60BFB530, srcCal
Sample Display The following output shows the DSP timestamp and the router timestamp for each event and, for SIG_STATUS, the state value shows the state of the ABCD bits in the signaling message. This sample shows a call coming in on a foreign exchange office (FXO) interface. The router waits for ringing to terminate before accepting the call.
Syntax Description slot-number Slot number in the router where the VIC is installed. Valid entries are from 0 to 2, depending on the slot where it has been installed. port Voice port. Valid entries are 0 or 1. Usage Guidelines Use the debug vpm port command to limit the debug output to a particular port. The debug output can be quite voluminous for a single port. A six-port chassis might create problems. Use this debug command with any or all of the other debug modes.
Sample Display The following output shows that a ring is detected and that the router waits for the ringing to stop before accepting the call: router# debug vpm signal ssm_process_event: [1/0, ssm_process_event: [1/0, ssm_process_event: [1/0, ssm_process_event: [1/0, 0.2, 0.7, 0.3, 0.3, 15] fxols_onhook_ringing 19] fxols_ringing_not 6] 19] fxols_offhook_clear The following output shows that the call is connected: ssm_process_event: [1/0, 0.3, 4] fxols_offhook_proc ssm_process_event: [1/0, 0.
The following output shows that the higher layer code accepts the call, requests addressing information, and starts DTMF and dial-pulse collection. This also shows that the digit timer is started. vcsm_process_event: [1/0, 0.
The following output shows the voice quality statistics collected periodically: vcsm_process_event: [1/0, 0.13, 17] dsp_get_rx_stats: [1/0] packet_len=12 channel_id=1 packet_id=87 reset_flag=0 vcsm_process_event: [1/0, 0.13, 28] vcsm_process_event: [1/0, 0.13, 29] vcsm_process_event: [1/0, 0.13, 32] vcsm_process_event: [1/0, 0.13, 17] dsp_get_rx_stats: [1/0] packet_len=12 channel_id=1 packet_id=87 reset_flag=0 vcsm_process_event: [1/0, 0.13, 28] vcsm_process_event: [1/0, 0.
debug vtsp dsp Use the debug vtsp dsp EXEC command to show messages from the digital signal processor (DSP) on the V.Fast Class (VFC) modem to the router. Use the no form of this command to disable debugging output. [no] debug vtsp dsp Usage Guidelines The debug vtsp dsp command shows messages from the DSP on the VFC to the router; this command is useful if you suspect that the VFC is not functional. It is a simple way to check if the VFC is responding to off-hook indications.
Sample Display The following example shows sample output from the debug vtsp error command, in which a dialed number is not reachable because it is not configured. router# deb vtsp error Voice telephony call control error debugging is on router# *Mar 1 00:21:48.698:cc_api_call_setup_ind (vdbPtr=0x1575AB0, callInfo={called=,called_oct3=0x81,calling=9999,calling_oct3=0x0,called_oct3a=0x0, fdest=0 peer_tag=1},callID=0x15896A4) *Mar 1 00:21:48.698:cc_api_call_setup_ind type 3 , prot 0 *Mar 1 00:21:48.
debug vtsp port To observe the behavior of the VTSP state machine on a specific voice port, use the debug vtsp port command. Use the no form of the command to turn off the debug function. For Cisco 1700 series with analog voice ports: debug vtsp port slot/port no debug vtsp port slot/port Sytnax Description For the Cisco 1700 series with analog voice ports: slot/port Debugs the analog voice port you specify with the slot/port designation.
Sample Display The following example shows sample output from the debug vtsp port 0/1 and debug vtsp all commands: router# debug vtsp port 0/1 21:59:14: vtsp_tsp_call_setup_ind (sdb=0x816CCA34, tdm_info=0x0, tsp_info=0x816CC600, calling_number= calling_oct3 = 0x0, called_number= called_oct3 = 0x81, oct3a=0x0): peer_tag=201 21:59:14: : ev.clg.clir is 0 ev.clg.clid_transparent is 0 ev.clg.null_orig_clg is 1 ev.clg.
21:59:16: Final pcn:, poa:, dial_string: 21:59:16: vtsp_get_dialpeer_tag: tag = 221 21:59:16: vtsp_get_dialpeer_tag: tag = 221 21:59:16: vtsp:[0/1:5505, S_PROCEEDING, E_CC_PROGRESS] 21:59:16: act_progress 21:59:16: vtsp_timer_stop: 7915625 21:59:16: vtsp:[0/1:5505, S_PROCEEDING, E_CC_BRIDGE] 21:59:16: act_bridge 21:59:16: vtsp_tdm_hpm_bridge 21:59:16: vtsp_tdm_hpm_bridge: cdb allow_tdm_hairpin = FALSE, dst_cdb_ptr allow_tdm_hairpin = TRUE 21:59:16: vtsp:[0/1:5505, S_PROCEEDING, E_CC_CAPS_IND] 21:59:16: act_
21:59:38: vtsp:[0/1:5505, S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN] 21:59:38: act_report_digit_begin 21:59:38: vtsp:[0/1:5505, S_CONNECT, E_DSP_DTMF_DIGIT] 21:59:38: act_report_digit_end 21:59:38: vtsp_timer_stop: 7917856 21:59:38: vtsp_timer: 7917856 21:59:39: vtsp:[0/1:5505, S_CONNECT, E_TSP_DISCONNECT_IND] 21:59:39: act_generate_disc 21:59:39: vtsp_ring_noan_timer_stop: 7917977 21:59:39: vtsp_timer_stop: 7917977 21:59:39: vtsp_pcm_tone_detect_timer_stop: 7917977 21:59:39: vtsp:[0/1:5505, S_CONNECT, E_CC_BRIDGE_
Sample Display The following output shows that the call has been accepted and that the system is now checking for incoming dial-peer matches: router# debug vtsp session *Nov 30 00:46:19.535: vtsp_tsp_call_accept_check (sdb=0x60CD4C58, calling_number=408 called_number=1): peer_tag=0 *Nov 30 00:46:19.
The following output shows that the call proceeded and that DTMF was disabled: *Nov 30 00:46:23.663: vtsp_process_event: [0:D:12, 0.4, 15] act_dcollect_proc *Nov 30 00:46:23.663: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:23.
The following output shows that the DSP channel was closed and released: *Nov 30 00:46:30.287: vtsp_process_event: [0:D:12, 0.18, 6] act_wrelease_release *Nov 30 00:46:30.287: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:30.287: dsp_idle_mode: [0:D:12] packet_len=8 channel_id=8737 packet_id=68 *Nov 30 00:46:30.287: dsp_close_voice_channel: [0:D:12] packet_len=8 channel_id=8737 packet_id=75 *Nov 30 00:46:30.287: vtsp_process_event: [0:D:12, 0.
debug vtsp tone To display debug messages showing the types of tones generated by the VoIP gateway, use the debug vtsp tone command. To disable the debug messages, use the no form of this command. [no] debug vtsp tone Sample Display The following example shows that a ringback tone was generated by the VoIP gateway: Router# debug vtsp tone *Jan 1 16:33:52.395:act_alert:Tone Ring Back generated in direction Network *Jan 1 16:33:52.
Sample Display The following example shows sample output from the debug vtsp vofr subframe command: router# debug vtsp vofr subframe 2 vtsp VoFR subframe debugging is enabled for payload 2 *Mar 6 18:21:17.413:VoFR frame received from Network AA AA AA *Mar 6 18:21:17.449:VoFR frame received from DSP (18 AA *Mar 6 18:21:23.969:VoFR frame received from Network AA AA AA *Mar 6 18:21:24.
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6 Routing Between Virtual LANs Overview This chapter provides an overview of virtual LANs (VLANs). It describes the encapsulation protocols used for routing between VLANs and provides some basic information about designing VLANs. This chapter describes VLANs.
• Security • Broadcast Control • Performance • Network Management • Communication Between VLANs LAN Segmentation VLANs allow logical network topologies to overlay the physical switched infrastructure such that any arbitrary collection of LAN ports can be combined into an autonomous user group or community of interest. The technology logically segments the network into separate Layer 2 broadcast domains whereby packets are switched between ports designated to be within the same VLAN.
Security VLANs also improve security by isolating groups. High-security users can be grouped into a VLAN, possible on the same physical segment, and no users outside that VLAN can communicate with them. Broadcast Control Just as switches isolate collision domains for attached hosts and only forward appropriate traffic out a particular port, VLANs provide complete isolation between VLANs. A VLAN is a bridging domain and all broadcast and multicast traffic is contained within it.
strips header and forwards the frame to interfaces that match the VLAN color. If you are using a Cisco network management product such as VlanDirector, you can actually color code the VLANs and monitor VLAN graphically. Why Implement VLANs? Network managers can group logically networks that span all major topologies, including high-speed technologies such as, ATM, FDDI, and Fast Ethernet.
• Sharing resources between VLANs • Load Balancing • Redundant Links • Addressing • Segmenting Networks with VLANs Segmenting the network into broadcast groups improves network security. Use router access lists based on station addresses, application types, and protocol types. • Routers and their Role in Switched Networks In switched networks, routers perform broadcast management, route processing and distribution, and provide communications between VLANs.
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7 Configuring Routing Between VLANs with IEEE 802.1Q Encapsulation This chapter describes the required and optional tasks for configuring routing between VLANs with IEEE 802.1Q encapsulation. For a complete description of VLAN commands used in this chapter, refer to the “Cisco IOS Switching Commands” chapter in the Cisco IOS Switching Services Command Reference. For documentation of other commands that appear in this chapter, you can use the command reference master index or search online. The IEEE 802.
To route AppleTalk over IEEE 802.1Q between VLANs, you need to customize the subinterface to create the environment in which it will be used. Perform these tasks in the order in which they appear: • Enabling AppleTalk Routing • Defining the VLAN Encapsulation Format • Configuring AppleTalk on the Subinterface Enabling AppleTalk Routing To enable AppleTalk routing on IEEE 802.
Configuring IP Routing over IEEE 802.1Q IP routing over IEEE 802.1Q extends IP routing capabilities to include support for routing IP frame types in VLAN configurations using the IEEE 802.1Q encapsulation. To route IP over IEEE 802.1Q between VLANs, you need to customize the subinterface to create the environment in which it will be used.
Configuring IPX Routing over IEEE 802.1Q IPX Routing over IEEE 802.1Q VLANs extends Novell NetWare routing capabilities to include support for routing Novell Ethernet_802.3 encapsulation frame types in VLAN configurations. Users with Novell NetWare environments can configure Novell Ethernet_802.3 encapsulation frames to be routed using IEEE 802.1Q encapsulation across VLAN boundaries. To configure Cisco IOS software on a router with connected VLANs to exchange IPX Novell Ethernet_802.
IEEE 802.1Q Encapsulation Configuration Examples This section provides configuration examples for each of the protocols described in this feature guide. It includes these examples: • Configuring AppleTalk over IEEE 802.1Q Example • Configuring IP Routing over IEEE 802.1Q Example • Configuring IPX Routing over IEEE 802.1Q Example Configuring AppleTalk over IEEE 802.1Q Example This configuration example shows AppleTalk being routed on VLAN 100. ! appletalk routing ! interface fastethernet 0/0.
VLAN Commands This section provides an alphabetical listing of all the VLAN commands that are new or specific to the Cisco 1751 router. All other commands used with this feature are documented in the Cisco IOS Release 12.1T command reference documents. clear vlan statistics To remove virtual LAN statistics from any statically or system configured entries, use the clear vlan statistics privileged EXEC command. clear vlan statistics Syntax Description This command has no arguments or keywords.
Example The following is sample output from the debug vlan packet output. Router# debug vlan packet Virtual LAN packet information debugging is on encapsulation dot1q To enable IEEE 802.1Q encapsulation of traffic on a specified subinterface in virtual LANs, use the encapsulation dot1q command in subinterface configuration mode. IEEE 802.1Q is a standard protocol for interconnecting multiple switches and routers and for defining VLAN topologies.
Command Mode Privileged EXEC Example The following is sample output from the show vlans command: 1751_2# show vlans Virtual LAN ID:1 (IEEE 802.1Q Encapsulation) vLAN Trunk Interface: FastEthernet0/0 This is configured as native Vlan for the following interface(s): FastEthernet0/0 Protocols Configured: Address: Received: Transmitted: Virtual LAN ID:100 (IEEE 802.1Q Encapsulation) vLAN Trunk Interface: FastEthernet0/0.100 Protocols Configured: IP Address: 100.0.0.
G L O S S A R Y A 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. ADPCM Adaptive differential pulse code modulation. Process by which analog voice samples are encoded into high-quality digital signals.
D DSP Digital signal processor. DSP segments the voice signal into frames and stores in voice packets. DTMF Dual tone multifrequency. Use of two simultaneous voice-band tones for dialing (such as touch tone). E E.164 International public telecommunications numbering plan. A standard set by ITU-T that addresses telephone numbers. E&M E&M interface uses a RJ-48 telephone cable to connect remote calls from an IP network to PBX trunk lines (tie lines) for local distribution.
P PBX Private branch exchange. Privately-owned central switching office. PCM Pulse code modulation. Transmission of analog information in digital form through sampling and encoding the samples with a fixed number of bits. PLAR Private line auto ringdown. PLAR connection associates a peer directly with an interface. 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.
T Service that provides 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. Trunk U User Datagram Protocol. UDP is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery, requiring that error processing and retransmission be handled by other protocols.
I N D E X Layer 2 6-2 A management, in VLANs 6-5 accounting per VLAN 6-3 Quality of Service (QoS) 6-3 C acc-qos command 4-4 call leg 2-9 addressing, in VLANs 6-4 CELP CODEC 1-3 ADPCM CODEC 1-3 central office (CO) 1-6 analog signals 1-3 CIR 2-24 answer-address command 4-5 Cisco IOS software documentation xi API 5-2 clear vlan statistics command 7-6 appletalk cable-range command 7-2 CODEC appletalk routing eigrp command 7-2 applied 1-2 appletalk zone command 7-2 command 4-6 audience xi
Index tasks 2-2 turning off 5-2 configuring using in a Telnet session 5-2 CODEC and VAD 2-23 when to use 5-1 custom queuing 2-7 debug vlan packet command 7-6 dial peers 2-9 debug voip ccapi error command 5-2 Frame Relay for VoIP 2-24 debug voip ccapi inout command 5-2 IP networks for real-time voice traffic 2-2 debug vpm all command 5-5 Multilink PPP interleaving 2-4 debug vpm dsp command 5-5 number expansion 2-8 debug vpm port command 5-6 POTS dial peer 2-12 debug vpm signal command 5-7
Index dial-peer voice command 4-13 Echo 1-5 dial-type command 4-14 echo-cancel coverage command 4-15 digital signal processor echo-cancel enable command 4-16 see DSP EEPROM 4-43 digital signals 1-3 encapsulation dot1q command 7-7 DLCI 2-24 examples DNS 2-26, 4-33 Frame Relay for VoIP 2-25 documentation VoIP configuration 3-1 CD ROM xi domain exit command 4-14 expect-factor command 4-17 bridging 6-1 broadcast 6-1 DSP F debug vpm dsp command 5-5 Fancy Queuing 2-2 defined 1-1 fax-rate c
Index G J ground start signaling 1-6, 4-57 jitter 1-5 H L H.
Index management 6-3 P VlanDirector 6-3 performance 6-4 packets, VLANs 7-6 scalability 6-4 PCM CODEC 1-3 security 6-4 performance 6-3, 6-4 services PLAR connection 4-8 accounting 6-3 port command 4-27 quality of service (QoS) 6-3 POTS dial peer security filtering 6-3 topology 6-4 configuring 2-12 described 2-10 networks, switched 6-5 prefix command 4-28 non-linear command 4-24 PVC 2-24 North American Numbering Plan 1-2 number expansion command 2-8 configuring 2-9 described 2-8 table 2-8
Index RTP header compression 2-6 S weighted fair queuing 2-7 scalability, in VLANs 6-4 security 6-4 R filtering 6-3 Random Early Detection 2-2 VLANs 6-2 segmentation 6-1, 6-2 RED see Random Early Detection with VLANs 6-5 redundancy in VLANs 6-4 session protocol command 4-32 req-qos command 4-29 session target command 4-32 resources, sharing between VLANs 6-4 session target dns command 4-33 ring frequency command 4-30 session target loopback command 4-33 ring number command 4-31 show call
Index trap message, generating 2-23 U trap operation, enabling 4-59 snmp enable peer-trap poor-qov command 4-58 UDP 1-2, 2-6 snmp-server enable traps command 4-59 snmp-server host command 4-59 snmp trap link-status command 4-60 V VAD configuring 2-24 T described 2-23 timeouts initial command 4-61 effect on comfort-noise command 4-8 timeouts interdigit command 4-62 effect on music-threshold command 4-23 timing command 4-63 vad command 4-67 traffic VFC modem 5-11 broadcast 6-3 VIC controlli
Index identifier 6-3 Frame Relay, configuring for 2-24 isolation between 6-3 Microsoft NetMeeting, configuring for 2-26 LAN segmentation 6-5 voice-port command 4-67 load balancing 6-4 voice ports monitoring 7-7 commands 4-3 network E&M changes 6-4 configuring 2-18 design 6-4 described 2-15 management 6-3 fine-tuning commands 2-20 performance 6-3 troubleshooting tips 2-20 performance 6-4 redundancy in 6-4 verifying 2-19 FXS/FXO routers in 6-5 configuring 2-15 routing between 6-4 des
Index WRED see Weighted Random Early Detection Cisco 1751 Router Software Configuration Guide OL-1070-01 ix
Index Cisco 1751 Router Software Configuration Guide x OL-1070-01
