Voice over IP for the Cisco AS5300 Feature Summary Voice over IP (VoIP) enables a Cisco AS5300 access server to carry voice traffic (for example, telephone calls and faxes) over an IP network. VoIP is primarily a software feature; however, to use this feature on the Cisco AS5300, you must install a VoIP feature card (VFC). Each VFC can hold up to five digital signal processor modules (DSPMs).
Feature Summary Figure 2 VoIP Used as a PSTN Gateway for Internet Telephone Traffic 408 526-4000 408 526-4001 408 526-4002 PSTN 408 526-4003 310 520-1001 310 520-1002 Central office 310 520-1000 310 520-1003 Voice port 1/0/0 Cisco AS5300 10.1.1.1 10.1.1.2 Cisco 3640 10352 IP cloud How VoIP Processes a Telephone Call Before configuring VoIP on your Cisco AS5300, it helps to understand what happens at an application level when you place a call using VoIP.
Benefits 7 Any call-progress indications (or other signals that can be carried in-band) are cut through the voice path as soon as an end-to-end audio channel is established. Signalling that can be detected by the voice 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.
Platforms 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 queueing scheme where the first calls received are the first calls processed. ISDN—Integrated Services Digital Network. ISDN is a communications protocol, offered by telephone companies, that permits telephone networks to carry data, voice, and other traffic.
List of Terms Prerequisites Before you can configure your Cisco AS5300 to use Voice over IP, you must first do the following: • 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 Cisco IOS 12.0 Network Protocols Configuration Guide, Part 1.
Configuration Tasks This feature supports the following RFCs: • RFC 1889—RTP: A Transport Protocol for Real-Time Applications, January 1996; H. Schulzrinne, GMD Fokus; S. Casner, Precept Software, Inc; R. Frederick, Xerox Palo Alto Research Centre; V. Jacobson, Lawrence Berkeley National Laboratory • RFC 1890—RTP Profile for Audio and Video Conferences with Minimal Control, January 1996; H. Schulzrinne, GMD Fokus • RFC 2127—ISDN Management Information Base using SMIv2, March 1997; G.
Configure IP Networks for Real-Time Voice Traffic Depending on the topology of your network or the resources used in your network, you might need to perform the following additional tasks: • • Distinguish Voice and Modem Calls on the Cisco AS5300 Optimize Dial Peer and Network Interface Configurations — Configure IP Precedence for Dial Peers — Configure RSVP for Dial Peers — Configure CODEC and VAD for Dial Peers • Configure Voice over IP for Microsoft NetMeeting Voice over IP for the Cisco AS5300 als
Configuration Tasks • Queueing 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.
Configure RTP Header Compression 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 perform the following tasks: • Configure the dialer interface or virtual template, as defined in the relevant chapters of the Cisco IOS Release 12.0 Dial Solutions Configuration Guide.
Configuration Tasks 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 substantial 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.
Configure Custom Queueing 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] Enables 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.
Configuration Tasks In general, weighted fair queueing is used in conjunction with Multilink PPP with interleaving and RSVP or IP Precedence to ensure that voice packet delivery. Use weighted fair queueing 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 queueing, refer to the Cisco IOS Release 12.0 Quality of Service Solutions Configuration Guide.
Frame Relay for Voice over IP Configuration Example 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.
Configuration Tasks For more information about Frame Relay, refer to the Cisco IOS Release 12.0 Wide-Area Networking Configuration Guide. Configure Voice Ports When an interface on the Cisco AS5300 is carrying voice data, it is referred to as a voice port.
Configure E1 R2 Voice Ports Step Command Purpose 10 pri-group timeslots range Configures ISDN PRI. 11 interface Serial0:23 Configures the IDSN D channel for the first ISDN PRI line. (The serial interface is the D channel.) 12 ip address ip-address Specifies an IP address for the interface. 13 isdn incoming-voice {voice | modem} Enables incoming ISDN voice calls. 14 interface Serial1:23 Configures the IDSN D channel for the second ISDN PRI line.
Configuration Tasks defines R2, but a number of countries and geographic regions implement R2 in entirely different ways. Cisco Systems addresses this lack of standards by supporting many localized implementations of R2 signalling in its Cisco IOS software. Cisco Systems’ E1 R2 signalling default is ITU, which supports the technology used in the following countries: Denmark, Finland, Germany, Russia (ITU variant), Hong Kong (ITU variant), and South Africa (ITU variant).
Configure E1 R2 Voice Ports Of the local variants listed above, the following local variants have been verified: • • • • • • Argentina Brazil China Mexico (Telmax) Singapore Thailand R2 signalling is channelized E1 signalling used in Europe, Asia, and South America. It is equivalent to channelized T1 signalling in North America. There are two types of R2 signalling: line signalling and interregister signalling. R2 line signalling includes R2 digital, R2 analog, and R2 pulse.
Configuration Tasks Step Command Purpose 4 country name use-defaults Specifies the local country, region, or corporation specification to use with R2 signalling. Replace the name variable with one of the supported country names. Cisco strongly recommends that you include the use-defaults option, which engages the default settings for a specific country. The default setting for all countries is ITU. See the cas-custom command in the Cisco IOS Release 12.
Configure E1 R2 Voice Ports Verify E1 R2 Signalling Configuration To verify the E1 R2 signalling configuration: • Type the show controller e1 command to view the status for all controllers, or type the show controller e1 number command to view the status for a particular controller. Make sure the status indicates the controller is up (line 2 in the following example) and no alarms (line 4 in the following example) or errors (lines 9 and 10 in the following example) have been reported.
Configuration Tasks 8 9 10 11 12 13 14 15 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 1 1 1 0 0 1 1 1 0 1 1 0 1 1 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 0 1 1 1 1 0 1 1 1 1 1 1 1 1 1 1 1 1 1 1 0 0 1 0 1 1 1 0 0 1 1 1 0 1 1 0 1 1 1 0 0 0 0 0 0 0 1 0 0 0 1 0 0 0 0 1 0 0 1 0 0 0 1 1 1 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 1 1 1 0 1 1 1 1 1 1 1 1 1 1 1 1 1 1 0 1 1 0 Tips If the connection does not come up, check
Configure T1 CAS Voice Ports Configure T1 CAS Voice Ports CAS is the transmission of signalling information within the voice channel. Various types of CAS signalling are available in the T1 world. The most common forms of CAS signalling are loop-start, ground-start, and E&M. The main disadvantage of CAS signalling is its use of user bandwidth to perform signalling functions.
Configuration Tasks — E&M Immediate Start In the Immediate Start protocol, the originating side does not wait for a wink before sending addressing information. After receiving addressing digits, the terminating side then goes off-hook for the duration of the call. The originating endpoint maintains off-hook for the duration of the call. • Ground Start / FXS—Ground Start signalling was developed to aid in resolving glare when two sides of a connection tried to go off-hook at the same time.
Configure T1 CAS Voice Ports Configure T1 CAS for Voice over IP To configure T1 CAS for Voice over IP on the Cisco AS5300, use the following commands beginning in privileged EXEC mode: Step Command Purpose 1 configure terminal Enters global configuration mode. 2 controller t1 number Enters controller configuration mode to configure your controller port. The controller ports are labeled 0 through 3 on the Quad T1/PRI and E1/PRI cards.
Configuration Tasks Step Command Purpose 11 cas-group channel timeslots range type signal Configures all channels for E&M, FXS, and SAS analog signalling. Enter 1-24 for T1. If E1, enter 1-31. Signalling types include e&m-fgb, e&m-fgd, e&m-immediate-start, fxs-ground-start, fxs-loop-start, sas-ground-start, and sas-loop-start. You must use the same type of signalling that your central office uses. For E1 using the Anadigicom converter, use cas e&m-fgb signalling.
Configure Number Expansion Verify T1 CAS Configuration To verify your controller is up and running and no alarms have been reported, perform the following task: • Enter the show controller t1 or show controller e1 command and specify the port number. 5300# show controller t1 2 T1 2 is up. No alarms detected. Version info of slot 0: HW: 2, Firmware: 16, PLD Rev: 0 Manufacture Cookie Info: EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x42, Board Hardware Version 1.
Configuration Tasks Figure 4 Sample Voice over IP Network 729 411-5002 729 411-5003 408 555-1001 729 411-5001 Cisco AS5300 Access Voice port Server 1 0:D WAN 408 555-2001 10.1.1.1 729 411-5004 T1 ISDN PRI Voice port 0:D IP cloud WAN 10.1.1.2 Cisco AS5300 Access Server 2 10351 1:D T1 ISDN PRI 408 555-3001 Table 1 shows the number expansion table for this scenario. Table 1 Sample Number Expansion Table Extension Destination Pattern Num-Exp Command Entry 1... 408555.... num-exp 1...
Configure Dial Peers 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 learn how a telephone number maps to a dial peer. 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 5 and Figure 6.
Configuration Tasks VoIP—VoIP dial peers describe the line characteristics usually associated with a packet network connection (in the case of VoIP, this is an IP network). VoIP peers define the line characteristics between VoIP devices—the routers and access servers carrying voice traffic in this voice network. 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 access server’s perspective.
Create a Peer Configuration Table the destination pattern this way means that router 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 in this document. Figure 8 shows how to complete the end-to-end call between dial peer 1 and dial peer 4. Outgoing Calls from the Perspective of POTS Dial Peer 2 Destination Source IP cloud Destination router Voice port 1/0/0 Source router 10.1.1.2 10.1.2.
Configuration Tasks Figure 9 Sample VoIP Network 729 411-5002 729 411-5003 408 555-1001 729 411-5001 408 555-2001 Cisco AS5300 Access Voice port Server 1 0:D WAN 10.1.1.1 729 411-5004 T1 ISDN PRI Voice port 0:D IP cloud WAN 10.1.1.2 Cisco AS5300 Access Server 2 10351 1:D T1 ISDN PRI 408 555-3001 Table 2 shows the peer configuration table for the example illustrated in Figure 4.
Configure POTS Peers To enter the dial peer configuration mode (and select POTS as the method of voice-related encapsulation), use the following commands in the global configuration mode: Command Purpose dial-peer voice number pots Enters 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.
Configuration Tasks PBX Incoming call leg Incoming and Outgoing POTS Call Legs AS5300 IP cloud AS5300 PBX Outgoing call leg 10369 Figure 10 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.
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 Enters the dial peer configuration mode to configure a POTS peer. 2 direct-inward-dial Specifies direct inward dial for this POTS peer. Note Direct inward dial is configured for the calling POTS dial peer.
Configuration Tasks Verify the Dial Peer 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 the dial peer to which a particular number (destination pattern) resolves.
Optimize Dial Peer and Network Interface Configurations If there is no called number information provided within the call setup, call classification is handled as follows: If there is a modem available in the system-default modem pool handle the call by a modem from this pool. Else handle the call as a voice call (either ise the voice dial peer assigned to the interface over which the call has arrived or use the default dial peer 0).
Configuration Tasks 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 QoS for a selected VoIP peer, use the following commands beginning in global configuration mode: Step Command Purpose 1 dial-peer voice number voip Enters the dial peer configuration mode to configure a VoIP peer.
Configure Voice over IP for Microsoft NetMeeting Configure CODEC for a VoIP Dial Peer 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 Enters the dial peer configuration mode to configure a VoIP peer. 2 codec [g711alaw | g711ulaw | g729r8] Specifies the desired voice coder rate of speech.
Configuration Tasks 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 CODEC—G711 U Law or G711 A Law Configure Microsoft NetMeeting for Voice Over IP To configure NetMeeting to work with Voice over IP, perform the following steps in the order given: 1 From the Tools menu in the NetMeeting application, select Options.
Download VCWare Table 3 VFC Firmware Extensions Firmware Filenames Description VCWare vcw-vfc-* Latest version of VCWare stores in Flash memory, including: • Datapath engine • Message dispatcher • DSP manager • VC manager • Process scheduler DSPWare btl-vfc-* DSP bootloader cor-vfc-* Core operating system and initialization bas-vfc-* Base voice cdc-*-* Voice CODEC files fax-vfc-* Fax relay files DSPWare is stored as a compressed file within VCWare; you must unbundle VCWare to install DSP
Configuration Tasks Determine the Number of VFCs To determine the number of installed VFCs and their location, use the following commands in privileged EXEC mode: Command Purpose show vfc slot directory Determines the number of installed VFCs and their location. For each VFC identified and located, perform the tasks described described in the following sections to upgrade system software on that VFC.
Copy Flash Files to the VFC Note If the VFC ROM is version 1.1, the image name must end in “.VCW.” If the VFC ROM is version 1.2, the image name must start with “vcv-.” Download Software (ROM Monitor Mode) To download VFC software to the VFC while the VFC is in ROM Monitor mode, perform the following tasks, beginning in privileged EXEC mode: Step Command Purpose 1 clear vfc slot purge Erase the VFC Flash memory.
Configuration Tasks Download VCWare to the VFC from a TFTP Server To download the latest version of VCWare from a TFTP server, make sure that the file is stored on the TFTP server. If you have a copy of the current version of VCWare on disk, you must store that image on a TFTP server before you can download the file to VFC memory.
Add CODECs to the Capability List Under most circumstances, these default files should be sufficient. If you need to, you can add an additional file (from those stored in VFC Flash memory) to the default file list or replace an existing file from the default file list. When you add a specific file to the default file list, it replaces the existing default for that extension type.
Configuration Tasks To delete a file from VFC Flash memory, use the following command in privileged EXEC mode: Command Purpose delete file-name vfc slot Deletes a specific file from the Flash memory on the VFC. Erase the VFC Flash Memory When you upgrade to a later version of VCWare, the new files are stored in VFC Flash, along with those already stored in VFC Flash memory—the new files do not overwrite existing files.
Linking PBX Users to a T1 ISDN PRI Interface Example Configuration Examples This section provides sample configurations for the following scenarios: • • • Linking PBX Users to a T1 ISDN PRI Interface Example Configuring Voice over IP for E1 R2 Signalling Example Configuring Voice over IP for T1-CAS Example These configuration examples should give you a starting point in your configuration process.
Configuration Examples Configuration for San Jose Access Server The first part of this configuration example defines dial-in access, including configuring the T1 lines and the ISDN D-channel parameters. For more information about configuring ISDN PRI, refer to the “Configuring Channelized E1 and Channelized T1” chapter in the Cisco IOS Release 12.0 Dial Solutions Configuration Guide.
Configuration for RTP Access Server ! Configure POTS dial-peer 3 using the second T1 dial-peer voice 3 pots prefix 5 dest-pat 1408555.... port 1:D ! ! Configure VoIP dial-peer 4 dial-peer voice 4 voip dest-pat 17294115... session-target ipv4:10.1.1.2 Configuration for RTP Access Server The first part of this configuration example defines dial-in access, including configuring the T1 line and the ISDN D-channel parameters.
Configuration Examples Configuring Voice over IP for E1 R2 Signalling Example The following example configures R2 signalling and customizes R2 parameters on controller E1 2 of a Cisco AS5300. In most cases, the same R2 signalling type is configured on each E1 controller. ! Specify the E1 controller that you want to configure with R2 signalling. A controller ! informs the access server how to distribute or provision individual timeslots for a ! connected channelized E1 line.
Configuring Voice over IP for T1-CAS Example no ip directed-broadcast no ip mroute-cache load-interval 30 no cdp enable ! interface FastEthernet0 ip address 173.14.25.100 255.255.0.0 no ip directed-broadcast bandwidth 1000000 load-interval 30 duplex full hold-queue 75 in ! no ip classless ip route 223.255.254.253 255.255.255.
Configuration Examples cas-group 1 timeslots 1-24 type e&m-fgb dtmf dnis ! Configure each additional controller (there are four). In this example, the ! controller number is 1, instead of 0. The clock source is secondary, instead of ! primary. The cas-group is 2, instead of 1 controller t1 1 framing esf linecode b8Zs clock source line secondary cas-group 2 timeslots 1-24 type e&m-fgb ! Configure each additional controller.
