Voice over IP for the Cisco AS5800 Configuration Overview

VoIP on the Cisco AS5800 has the following primary applications:

• Toll bypass

• Remote PBX presence over WANs

• Unified voice/data trunking

• POTS-Internet telephony gateways

ANI---Answer Number Indication. The calling number (number of calling party).

ARQ---Admission request.

Data Link Connection Identifier (DLCI)---Frame Relay virtual circuit number corresponding to a particular destination. The DLCI is part of the Frame Relay header and is usually 10 bits long.

DNS---Domain name system is used to address translation to convert H.323 IDs, URLs, or e-mail IDs to IP addresses. DNS is also used to assist in the location of remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains.

DNIS---Dialed number identification service. The destination number.

DSP---Digital Signal Processor.

E.164---The international public telecommunications numbering plan. A standard set by ITU-T which addresses telephone numbers.

E&M---Ear and mouth RBS signalling

Endpoint---An H.323 terminal or gateway. An endpoint can call and be called. It generates and/or terminates the information stream.

Gatekeeper---A gatekeeper maintains a registry of devices in the multimedia network. The devices register with the gatekeeper at startup, and request admission to a call from the gatekeeper.

The gatekeeper is an H.323 entity on the LAN that provides address translation and control access to the LAN for H.323 terminals and gateways. The gatekeeper may provide other services to the H.323 terminals and gateways, such as bandwidth management and locating gateways.

Gateway---A gateway allows H.323 terminals to communicate with non-H.323 terminals by converting protocols. A gateway is the point at which a circuit-switched call is encoded and repackaged into IP packets.

An H.323 gateway is an endpoint on the LAN that provides real-time two-way communications between H.323 terminals on the LAN and other ITU-T terminals in the WAN, or to another H.323 gateway.

H.323---An International Telecommunication Union (ITU-T) standard that describes packet-based video, audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the conferencing system, and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual protocol.

H.323 RAS---Registration, admission, and status. The RAS signaling function performs registration, admissions, bandwidth changes, status and disengage procedures between the VoIP gateway and the gatekeeper.

HSRP---Hot Standby Routing Protocol. HSRP is a Cisco proprietary protocol which provides a redundancy mechanism when more than one router is connected to the same segment/subnet of an Ethernet/FDDI/Token Ring network.

ITU-T---Telecommunication standardization sector of ITU.

IVR---Integrated voice response. A software feature that allows the use of one of several interactive voice response scripts during the call processing functionality.

LEC---Local exchange carrier.

LRQ---Location request.

MCU---Multipoint control unit

Multicast---A process of transmitting PDUs from one source to many destinations. The actual mechanism (that is, IP multicast, multi-unicast, etc.) for this process may be different for LAN technologies.

Multipoint-unicast---A process of transferring PDUs (Protocol Data Units) where an endpoint sends more than one copy of a media stream to different endpoints. This may be necessary in networks which do not support multicast.

node---An H.323 entity that uses RAS to communicate with the gatekeeper. For example, an endpoint such as a terminal, proxy, or gateway.

PDU---Protocol Dara Units, used by bridges to transfer connectivity information.

RAS---Registration, Admission, and Status protocol. This is the protocol that is used between endpoints and the gatekeeper to perform management functions.

RBS---Robbed Bit Signaling

RRQ---Registration request.

Tool Command Language---TCL is an interpreted script language developed by Dr. John Ousterhout of the University of California, Berkeley, and is now developed and maintained by Sun Microsystems Laboratories.

SPI---Service provider interface.

TDM---Time division multiplexing. Technique in which information from multiple channels can be allocated bandwidth on a single wire based on preassigned time slots. Bandwidth is allocated to each channel regardless of whether the station has data to transmit.

VoIP---Voice over IP. The ability to carry normal telephone-style voice over an IP-based internet with POTS-like functionality, reliability, and voice quality. VoIP is a blanket term which generally refers to Cisco's standards based (H.323, etc.) approach to IP voice traffic.

VTSP---Voice telephony service provider.

Zone---A collection of all terminals (tx), gateways (GW), and Multipoint Control Units (MCU) managed by a single gatekeeper (GK). A Zone includes at least one terminal, and may or may not include gateways or MCUs. A Zone has only one gatekeeper. A Zone may be independent of LAN topology and may be comprised of multiple LAN segments which are connected using routes or other devices.

• Cisco AS5800 universal access servers

• Cisco AS5300 universal access servers

• Cisco 3600 series routers

The configuration procedure described in this document pertains to the Cisco AS5800.

• Be sure that you are running Cisco IOS Releases 12.0(4)XL that requires DSDWare Version 3.1.7

• 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 11.3 Network Protocols Configuration Guide, Part 1.

• Complete basic configuration for the AS5800. This includes, as a minimum, the following tasks:

  • Configure a host name and password for the AS5800

  • Configure the Fast Ethernet interface of your AS5800 so that it can be recognized as a device on the Ethernet LAN

  • Configure the AS5800 interfaces for ISDN PRI lines

  • Configure the ISDN D channels for each ISDN PRI line

  • Configure the AS5800 interfaces for T1 CAS lines

  • Configure the ISDN D channels for each T1 CAS PRI line

For more information about any of the these configuration tasks, refer to the Cisco AS5800 Universal Access Server Software Installation and Configuration Guide, which shipped with your Cisco AS5800 and is available on the document CD-ROM.

• Install the VFC into the appropriate slot of your Cisco AS5800 universal access server. Each VFC can hold up to 16 digital signal processor modules (DSPMs), enabling processing for up to 192 voice channels. For more information about the physical characteristics of the VFCs or DSPMs, or how to install them, refer to Installing Voice over IP Feature Cards in Cisco AS5800 Universal Access Servers, which shipped with your VFC and is available online.

• Complete your company's dial plan.

• Establish a working telephony network based on your company's dial plan.

• Integrate your dial plan and telephony network into your existing IP network topology. Merging your IP and telephony networks depends on your particular IP and telephony network topology. In general, we recommend the following suggestions:

  • Use canonical numbers wherever possible. It is important that you avoid situations where numbering systems are significantly different on different routers or access servers in your network.

  • Make routing and/or dialing transparent to the user. For example, avoid secondary dial tones from secondary switches, where possible.

  • Contact your PBX vendor for instructions about how to reconfigure the appropriate PBX interfaces.

QoS must be configured throughout your network---not just on the Cisco AS5800 devices running VoIP---to improve voice network performance. Not all QoS techniques are appropriate for all network routers. Edge routers and backbone routers in your network do not necessarily perform the same operations; the QoS tasks they perform might also differ. To configure your IP network for real-time voice traffic, you need to consider the functions of both edge and backbone routers in your network, then select the appropriate QoS tool or tools.

In general, edge routers perform the following QoS functions:

• Packet classification

• Admission control

• Bandwidth management

• Queuing

In general, backbone routers perform the following QoS functions:

• High-speed switching and transport

• Congestion management

• Queue management

Scalable QoS solutions require cooperative edge and backbone functions.

Although not required, you can use the custom queuing QoS tool to fine-tune your network for real-time voice traffic.

Each of these components is discussed in the following sections.

Custom Queuing and other methods for identifying high priority streams should be configured for these port ranges. For more information about custom queuing, refer to the "Managing System Performance" chapter in the Cisco IOS Release 11.3 Configuration Fundamentals Configuration Guide.

• Use the show voice port command to verify that the data configured is correct.

• If you have not configured your device to support direct inward dial, dial in to the router and see if you have dial tone.

• Enter DTMF digit. If the dial tone stops, you have two-way voice connectivity with the router.

The information included in this example needs to be configured on both Access Server 1 and Access Server 2.

For inbound call legs, a dial peer might be associated to the calling number or the port designation. Outbound call legs always have a dial peer associated with them. The destination pattern is used to identify the outbound dial peer. The call is associated with the outbound dial peer at setup time.

POTS peers associate a telephone number with a particular voice port so that incoming calls for that telephone number can be received and outgoing calls can be placed. VoIP peers point to specific devices (by associating destination telephone numbers with a specific IP address) so that incoming calls can be received and outgoing calls can be placed. Both POTS and VoIP peers are needed to establish Voice over IP connections.

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.) Use the following steps to configure the identified POTS peer.

Step	Command	Task
1	5800> enable password: 5800#	Enter enable mode. Enter the password. You have entered enable mode when the prompt changes 5800#.
2	5800# config term Enter configuration commands, one per line. End with CNTL/Z. 5800(config)#	Enter global configuration mode. You have entered global configuration mode when the prompt changes to 5800(config)#.
3	5800(config)#destination-pattern string	Define the telephone number associated with this POTS dial peer.
4	5800(config)#port shelf/slot/port:D	Associate this POTS dial peer with a specific logical dial interface.

For example, suppose there is a voice call whose E.164 called number is 1 310 767-2222. If you configure a destination-pattern of "1310767" and a prefix of "9," the router will strip out "1310767" from the E.164 telephone number, leaving the extension number of "2222." It will then append the prefix, "9," to the front of the remaining numbers, so that the actual numbers dialed is "9, 2222." The comma in this example means that the router will pause for one second between dialing the "9" and the "2" to allow for a secondary dial tone.

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 is identified, the call is forwarded through the next call leg to the destination.

There are cases where it might be necessary for the server to use the called-number (DNIS) to find a dial peer for the outgoing call leg---for example, if the switch connecting the call to the server has already collected the digits. DID enables the server to match the called-number with a dial peer and then directly place the outbound call. With DID, the server does not present a dial tone to the caller and does not collect digits; it forwards the call directly to the configured destination.

To use DID and incoming called-number, a dial peer must be associated with the incoming call leg. Before doing this, it helps if you understand the logic behind the algorithm used to associate the incoming call leg with the dial peer.

The algorithm used to associate incoming call legs with dial peers uses three inputs (which are derived from signaling and interface information associated with the call) and four defined dial peer elements. The three signaling inputs are:

• Called-number (DNIS)---Set of numbers representing the destination, which is derived from the ISDN setup message or CAS DNIS.

• Calling-number (ANI)---Set of numbers representing the origin, which is derived from the ISDN setup message or CAS DNIS.

• Voice port---The voice port carrying the call.

The four defined dial peer elements are:

• Destination pattern---A pattern representing the phone numbers to which the peer can connect.

• Answer address---A pattern representing the phone numbers from which the peer can connect.

• Incoming called-number---A pattern representing the phone numbers that associate an incoming call leg to a peer based on the called-number or DNIS.

• Port---The port through which calls to this peer are placed.

Using the elements, the algorithm is as follows:

For all peers where call type (VoIP versus POTS) match dial peer type:
if the type is matched, associate the called number with the incoming called-number
 else if the type is matched, associate calling-number with answer-address
 else if the type is matched, associate calling-number with destination-pattern
 else if the type is matched, associate voice port to port
 

This algorithm shows that if a value is not configured for answer-address, the origin address is used because, in most cases, the origin address and answer-address are the same. Use the following steps to configure DID for a particular POTS dial peer.

Step	Command	Task
1	5800> enable password: 5800#	Enter enable mode. Enter the password. You have entered enable mode when the prompt changes 5800#.
2	5800# config term Enter configuration commands, one per line. End with CNTL/Z. 5800(config)#	Enter global configuration mode. You have entered global configuration mode when the prompt changes to 5800(config)#.
3	5800(config)# dial-peer voice number pots	Enter the dial peer configuration mode to configure a POTS peer.
4	5800(config)# direct-inward-dial	Specify direct inward dial for this POTS peer.

Use the following steps to enter the dial peer configuration mode (and select VoIP as the method of voice-related encapsulation).

Step	Command	Task
1	5800> enable password: 5800#	Enter enable mode. Enter the password. You have entered enable mode when the prompt changes 5800#.
2	5800# config term Enter configuration commands, one per line. End with CNTL/Z. 5800(config)#	Enter global configuration mode. You have entered global configuration mode when the prompt changes to 5800(config)#.
3	5800(config)# dial-peer voice number voip	Enter the dial peer configuration mode to configure a VoIP peer.

The number value of the dial-peer voice voip command is a tag that uniquely identifies the dial peer. Use the following steps to configure the identified VoIP peer.

Step	Command	Task
1	5800> enable password: 5800#	Enter enable mode. Enter the password. You have entered enable mode when the prompt changes 5800#.
2	5800# config term Enter configuration commands, one per line. End with CNTL/Z. 5800(config)#	Enter global configuration mode. You have entered global configuration mode when the prompt changes to 5800(config)#.
3	5800(config)# destination-pattern string	Define the destination telephone number associated with this VoIP dial peer.
4	5800(config)# session-target {ipv4:destination-address | dns:host-name}	Specify a destination IP address for this dial peer.

• 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.

• Ping the associated IP address to confirm connectivity. If you cannot successfully ping your destination, refer to the chapter, "Configuring IP," in the Cisco IOS 11.3 Network Protocols Configuration Guide, Part 1.

• Use the show dial-peer voice command to verify that the operational status of the dial peer is up.

• Use the show dialplan number command on the local and remote routers to verify that the data is configured correctly on both.

• If you have configured number expansion, use the show num-exp command to check that the partial number on the local router maps to the correct full E.164 telephone number on the remote router.

• If you have configured a CODEC value, there can be a problem if both VoIP dial peers on either side of the connection have incompatible CODEC values. Make sure that both VoIP peers have been configured with the same CODEC value.

• Use the debug voip ccapi inout command to verify the output string the router dials is correct.

• Use the debug cch323 rtp command to check RTP packet transport.

• Use the debug cch323 h245 command to check logical channel negotiation.

• Use the debug cch323 h225 command to check the call setup.

To give real-time voice traffic precedence over other IP network traffic, perform the following tasks, starting in global configuration mode:

Step	Command	Task
1	5800> enable password: 5800#	Enter enable mode. Enter the password. You have entered enable mode when the prompt changes 5800#.
2	5800# config term Enter configuration commands, one per line. End with CNTL/Z. 5800(config)#	Enter global configuration mode. You have entered global configuration mode when the prompt changes to 5800(config)#.
3	5800(config)# dial-peer voice number voip	Enter the dial peer configuration mode to configure a VoIP peer.
4	5800(config)# ip precedence number	Select a precedence level for the voice traffic associated with that dial peer.

In IP Precedence, the numbers 1 through 5 identify classes for IP flows; the numbers 6 through 7 are used for network and backbone routing and updates.

For example, to ensure that voice traffic associated with VoIP dial peer 103 is given a higher priority than other IP network traffic, enter the following:

dial-peer voice 103 voip
 ip precedence 5
 

In this example, when an IP call leg is associated with VoIP dial peer 103, all packets transmitted to the IP network via this dial peer will have their precedence bits set to 5. If the networks receiving these packets have been configured to recognize precedence bits, the packets will be given priority over packets with a lower configured precedence value.

The default for the codec command is g729r8; normally the default configuration for this command is the most desirable. If, however, you are operating on a high bandwidth network and voice quality is of the highest importance, you should configure the codec command for g711alaw or ulaw. Using this value will result in better voice quality, but it will also require higher bandwidth requirements for voice.

For example, to specify a CODEC rate of G.711a-law for VoIP dial peer 108, enter the following:

Dial-peer voice 108 voip
 destination-pattern +1408528
 codec g711alaw
 session target ipv4:10.0.0.8

The default for the vad command is enabled; normally the default configuration for this command is the most desirable. If you are operating on a high bandwidth network and voice quality is of the highest importance, you should disable vad. Using this value will result in better voice quality, but it will also require higher bandwidth requirements for voice.

For example, to enable VAD for VoIP dial peer 108, enter the following:

Dial-peer voice 108 voip
 destination-pattern +1408528
 vad
 session target ipv4:10.0.0.8

Step 1 From the Tools menu in the NetMeeting application, select Options. NetMeeting will display the Options dialog box.

Step 2 Click the Audio tab.

Step 3 Click the "Calling a telephone using NetMeeting" check box.

Step 4 Enter the IP address of the Cisco AS5800 in the IP address field.

Step 5 Under General, click Advanced.

Step 6 Click the "Manually configured compression settings" check box.

Step 7 Select the CODEC value CCITT ulaw 8000Hz.

Step 8 Click the Up button until this CODEC value is at the top of the list.

Step 9 Click OK to exit.

Step 1 Click the Call icon from the NetMeeting application. Microsoft NetMeeting will open 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.

Step 4 Click Call to initiate a call to the Cisco AS5800 from Microsoft NetMeeting.