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Chapter 4 - Voice over IP (VoIP) Configurations

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Chapter 4 - Voice over IP (VoIP)
Chapter 4 - Voice over IP (VoIP) Voice over IP Configurations Configurations
Configurations
Overview Feedback

Introduction

Voice Handling

Quality of Service Support

H.323v2 Protocol

SGCP and MGCP Protocol Stack

H.323v2 Static Bridging Configuration

H.323v2 Static Routing Configuration

H.323v2 Dynamic Mapping Configuration

SGCP Configuration

MGCP Configuration

Voice over IP Configurations

This chapter provides an overview of Voice over IP (VoIP) operations on the Cisco uBR924 cable access router. It
also describes how to configure the Cisco uBR924 router for basic VoIP operation in both bridging and routing
modes. This chapter contains the following sections:

• Overview

• H.323v2 Static Bridging Configuration

• H.323v2 Static Routing Configuration

• H.323v2 Dynamic Mapping Configuration

• SGCP Configuration

• MGCP Configuration

Note The configurations shown in this chapter can be combined with most of the data-only configurations shown
in "Advanced Data-Only Configurations." All voice configurations assume that the CMTS and associated servers,
gateways, and gatekeepers have been configured accordingly.

Overview
When using a Cisco IOS image that contains voice support, the Cisco uBR924 cable access router supports
Voice over IP (VoIP), which transmits voice and fax calls over a TCP/IP network such as the Internet. Depending on
the services purchased from the cable service provider, subscribers can place and receive calls without using the
local telco exchange carrier.

The Cisco uBR924 router has two voice ports that support two simultaneous voice and fax calls from each
subscriber site, but multiple telephones and fax devices can be connected to each of the two VoIP telephone lines
(provided that the 5 REN limit for each telephone line is not exceeded). Telephones at each subscriber site must
support touch-tone dialing; rotary dialing is not supported. Special telephone features such as call waiting,
forwarding, and conferencing are supported only when using Cisco IOS images that support those features.

Note Fax devices—standard Group III and computer-based Group III machines up to 14,400 baud—are supported
in Cisco IOS Release 12.0(5)T and higher images that support VoIP. However, in general, fax/modem cards are
not supported over VoIP links. You must be using a Cisco IOS image that supports voice and have purchased the
appropriate feature license before being able to make voice calls using the Cisco uBR924 router.

Introduction

The Cisco uBR924 router uses packets to transmit and receive digitized voice over an IP network. Voice traffic is
supported in both the DOCSIS-bridging and routing modes.

Note When the router is acting in DOCSIS-bridging mode, a voice call originating from the router's Ethernet
interface cannot terminate on another device attached to that same Ethernet interface; it must terminate on a
device that is reached through the cable interface. The router must be operating in routing mode to allow calls to
both originate and terminate on the Ethernet interface.

Voice signals are packetized and transported in compliance with the following protocols:

• H.323v2—Second version of an International Telecommunications Union (ITU) standard that specifies call
signaling and control protocols for an IP data network. Supported on Cisco IOS Release 12.0(4)XI and higher
voice images.

• Simple Gateway Control Protocol (SGCP) Version 1.1—A signaling protocol under review by the Internet
Engineering Task Force (IETF). Supported on Cisco IOS Release 12.0(7)T and higher voice images.

• Media Gateway Control Protocol (MGCP) Version 0.1—A proposed IETF voice control protocol intended to
eventually supersede the existing SCGP 1.1 protocol. Supported on Cisco IOS Release 12.1(3)T and higher voice
images.

Note In Cisco IOS Release 12.1(3)T, the MGCP 0.1 and SGCP 1.1 protocols have been merged on the
Cisco uBR924 router so that the router can respond efficiently to either protocol. The MGCP and SGCP protocols
cannot be used if the H.323v2 protocol is used.

Figure 4-1 illustrates a broadband cable system that supports VoIP transmission.

Figure 4-1 Simplified VoIP Network

The CMTS at the headend routes IP telephony calls from the point of origination to the destination, transmitting
them along with other traffic (both voice and data). To route voice calls across the local IP network to a destination
on the Internet or the public switched telephone network (PSTN), the Cisco uBR924 router and CMTS deploy IP
telephony as a local-loop bypass service. One of the following routing methods is then used, depending on the
protocol being used:

• If using H.323v2, the Cisco uBR924 acts as the H.323v2 gateway that forwards the voice packets to the CMTS,
which then sends them to a telephony gatekeeper. The gatekeeper transmits the packets to their ultimate
destination.

• If using SGCP or MGCP, the Cisco uBR924 router acts as the residential gateway that forwards the voice
packets to the CMTS, which then connects to the external call agent (SGCP or MGCP) or media gateway controller
(MGCP). The call agent or controller determines how to transmit the call across the network to the trunking
gateway that will be its ultimate destination.

The gateway at the destination typically interconnects the IP network to the public switched telephone network
(PSTN) so that calls can be made to any phone, not just those that are part of the IP telephony network.

Voice calls are digitized, encoded, compressed, and packetized in an originating gateway; and then,
decompressed, decoded, and reassembled in the destination gateway. A server maintains subscriber profiles
and policy information. See the Cisco service provider voice documentation set if you have Cisco gatekeeper,
gateway, or other applicable products.

Caution In certain countries, the provisioning of voice telephony over the Internet or use of these products may be
prohibited and/or subject to laws, regulations or licenses, including requirements applicable to the use of the
products under telecommunications and other laws and regulations; customer must comply with all such
applicable laws in the country where the customer intends to use the product.

Voice Handling

With IP telephony, telephone calls can be delivered at rates as low as 8 kbps in a packet format using
compression algorithms. Depending on the software release used, the Cisco uBR924 cable access router
supports the following algorithms:

• G.711 A-Law—64000 bps PCM uncompressed encoding, using the A-Law standard used in most of the world
except for North America and a few other countries.

• G.711 Mu-Law—64000 bps PCM uncompressed encoding, using the Mu-Law standard used in North
America and a few other countries.

• G.729—8000 bps compressed CS-ACELP encoding (default for telephone calls).

Caution Because voice is delay-sensitive, a well-engineered network is critical. Fine-tuning your network to
adequately support VoIP typically involves a series of protocols and features geared to support QoS.

To achieve acceptable voice quality and reduce network bandwidth usage, several voice processing techniques
are used. Digital Signal Processors (DSPs) provide the stream-to-packet and packet-to-stream conversion, as
well as voice processing capabilities. Typical voice processing services include echo cancellation, voice
compression, Voice Activity Detection (VAD) or silence compression, and Dual Tone Multi-Frequency (DTMF) tone
detection and generation.

Quality of Service Support

Data traffic typically is sent only on a "best effort" basis, and if a packet is lost or delayed, it can be easily
retransmitted without significantly affecting the connection. Such delays and losses are unacceptable, however,
for real-time traffic such as voice calls.

For this reason, the CMTS and Cisco uBR924 router assign separate service identifiers (SIDs) for the voice and
data traffic flows. Each SID has a separate class of service (CoS) that determines how its traffic flow is handled,
allowing voice traffic to have a higher priority than the data traffic.

The CMTS and router can use different traffic shaping mechanisms to ensure that the higher priority voice traffic
always has the bandwidth it needs. This allows voice calls (and other real-time traffic) to share the same channel
as data traffic, without the quality of the voice calls being degraded by bursty data transmissions.

Note Separate CoS flows are available only when the router is connected to a CMTS that supports multiple
classes of service per router. In addition, the router's configuration file must enable multiple classes of service.

The DOCSIS 1.0 specification does not support multiple CoS flows, so this flow technique is not available when
the Cisco uBR924 router interoperates with a DOCSIS 1.0 CMTS. In this situation, voice and data traffic are both
transmitted on a "best effort" basis. This may cause poorer voice quality and lower data throughput when calls are
being made from the router's telephone ports.

The Cisco uBR924 router supports the following service classes:

• The first CoS in the router's configuration file is configured as the "Tiered Best Effort Type Class" and is the
default CoS for data traffic. The class has no minimum upstream rate specified for the channel.

This service class is assigned to the primary SID for the router. In addition to being used for data traffic, the router
uses this SID for all MAC message exchanges with the CMTS, as well as for SNMP management traffic.

All traffic using this SID is transmitted on a "best effort" basis, but data traffic within this class can be prioritized
into eight different priority levels; although all data traffic still has lower priority than the voice traffic, this allows
certain data traffic (such as MAC messages) to be given higher priority than other data traffic. The CMTS system
administrator defines the traffic priority levels and must include the traffic priority fields in the configuration file
downloaded to the Cisco uBR924.

• The second and third CoS are for the first and second voice ports, respectively, which are assigned to the
secondary SIDs used for the voice ports. If using a Cisco IOS image that supports dynamic multi-SID assignment,
these secondary SIDs are automatically created when a call is placed from one of the voice ports; when the call
terminates, the secondary SID associated with it is deleted. If the Cisco IOS image does not support multi-SIDs,
static SIDs are created for each of the voice ports during the power-on provisioning process, permanently
reserving the bandwidth needed for the voice traffic.

The CMTS system administrator typically configures these secondary classes of service so that they have higher
QoS classes for use by higher priority voice traffic. These classes should also have a minimum upstream data
rate specified for the channel to guarantee a specific amount of bandwidth for the corresponding traffic flows.
When static SIDs are used, that bandwidth is always reserved for voice calls; however, when dynamic multi-SID
assignment is used, that bandwidth is reserved only when the voice calls are active.

H.323v2 Protocol

In architectures using the VoIP H.323v2 protocol stack, the session application manages two call legs for each
call: a telephony leg managed by the voice telephony service provider and the VoIP leg managed by the cable
system operator—the VoIP service provider. Use of the H.323v2 protocol typically requires a dial plan and mapper
at the headend or other server location to map IP addresses to telephone numbers.

When both legs of the call have been setup, the session application creates a conference between them. The
opposite leg's transmit routine for voice packets is given to each provider. The CMTS router passes data to the
gateway and gatekeeper. The H.323v2 protocol stack provides signaling via H.225 and feature negotiation via
H.245.

Note For more information on using H.323v2, see the document H.323 Version 2 Support, available on CCO and
the Documentation CD-ROM.

To make and receive H.323 calls, the Cisco uBR924 router must be configured for the following:

• The IP address of the gateway for the destination dialed—In Cisco uBR924 IOS Release 12.0(4)XI or higher
interim builds, configure these IP addresses statically via the command-line interface (CLI) using voip dial peer
group commands. When running Cisco IOS Release 12.0(5)T or higher interim images on Cisco gatekeeper
products, the router obtains these addresses dynamically from the gatekeeper using the Registration, Admission,
and Status (RAS) protocol.

• The telephone numbers of the attached devices—In Cisco IOS Release 12.0(4)XI or higher interim builds, you
configure these IP addresses statically via the CLI pots port commands. When using Cisco Network Registrar
(CNR) version 3.0 or higher with the relay.tcl and setrouter.tcl scripts, and Cisco gatekeeper products in your
network running Cisco IOS Release 12.0(5)T or higher images, you can obtain these addresses dynamically from
CNR. The telephone numbers of attached devices are then sent in DHCP response messages. When the
Cisco uBR924 processes the DHCP response, it automatically creates the pots dial peer for each port, creates
the voip dial peer for the RAS target, and starts the H.323v2 RAS gateway support.

Note To support voice configurations involving Cisco gatekeeper products using RAS, Cisco IOS
Release 12.0(5)T or higher images with gatekeeper support are required. The headend must have IP multicast
enabled. The cable interface must be designated as the default for RAS to discover the gatekeeper. The
gatekeeper then resolves all dialed destinations sent to the RAS protocol.

SGCP and MGCP Protocol Stack

When using a Cisco IOS Release 12.0(5)T or higher image with voice support, the Cisco uBR924 router supports
the Simple Gateway Control Protocol (SGCP). When using a Cisco IOS Release 12.1(3)T or higher image with
voice support, the Cisco uBR924 router also supports the MGCP protocol, which is intended to eventually
supersede the SGCP protocol. Both MGCP and SGCP are signaling protocols that interact with a remote call
agent (CA) to provide call setup and teardown for VoIP calls.

Using the call agent, SGCP and MGCP communicate with the voice gateways, dynamically resolving and routing
calls. This creates a distributed system that enhances performance, reliability, and scalability while still appearing
as a single VoIP gateway to external clients.

The remote call agent also provides the signaling and feature negotiation that would otherwise be provided by the
Cisco uBR924 router when using the H.323v2 protocol. Similarly, the call agent also provides the mapping of IP
addresses to telephone numbers, eliminating the dial plan mapper and static configurations that are required on
the router when using the H.323v2 protocol.

The SGCP and MGCP protocols implement the gateway functionality using both trunk and residential gateways.
The Cisco uBR924 router functions in this mode as a residential gateway with two endpoints.

SGCP and MGCP can preserve Signaling System 7 (SS7) style call control information as well as additional
network information such as routing information and authentication, authorization, and accounting (AAA) security
information. SGCP and MGCP allow voice calls to originate and terminate on the Internet, as well as allowing one
end to terminate on the Internet and the other to terminate on a telephone on the PSTN.

Note The Cisco uBR924 cable access router supports both H.323 and SGCP/MGCP call control, but only one
method can be active at a time.

H.323v2 Static Bridging Configuration

When the Cisco uBR924 router is running in DOCSIS-bridging mode and using a Cisco IOS image with voice
support, it can route voice calls using an H.323v2 static dialing map. This requires the following minimum
configuration:

• Create a local dial peer for each voice port that will receive incoming calls. This requires configuring each
voice port on the router with the phone numbers for the devices attached to those voice ports. The Cisco uBR924
router uses these numbers to determine which voice port should receive the call. Typically, the complete phone
number or extension is specified for each port; when the Cisco uBR924 router receives an incoming call, all digits
in the number are matched and stripped off, and the voice port is connected to the call.

Note The voice ports on the Cisco uBR924 router support only FXS devices.

• Configure a remote dial peer for each possible destination for outgoing calls. This requires specifying the
phone number(s) for the destination devices. Use the following guidelines for what numbers to enter:

– For a single telephony device, such as a one-line phone or fax machine, enter the complete phone number or
extension.

– To direct a group of numbers to a specific destination—such as the extensions used on a remote PBX—enter
a pattern matching the prefix used for those lines; asterisks can be used to match any number of digits and a
period matches a single digit. For example, "572*" matches any phone numbers starting with 572 while "572."
matches the numbers 5720-5729.

You must also specify the IP address for the destination host that will deliver the call to the telephony device (or if
the destination device is an IP telephone, the IP address for that telephone). You can optionally specify an IP
precedence level for the type of service (ToS) bits in the IP header to signify that these voice packets should be
given higher priority in transit across the IP network.

If not being done by the CoS, you can also specify which coding/decoding (CODEC) algorithm should be used.

These functions are done using the dial-peer command, as shown in the following table:

Command Purpose

Step 1 To configure incoming calls Specify a unique id-numb er for this incoming
on voice port V1: dial-peer and enter dial-peer configuration
mode.
uBR924(config)# dial-peer
voice id-numb er pots

Step 2 uBR924(config-dial-peer)# Specify the telephone number(s) associated

destination-pattern digits with this voice port.

Step 3 uBR924(config-dial-peer)# Specify that voice port V1 is attached to this

port 0 telephony equipment.

Step 4 uBR924(config-dial-peer)# Optionally configure the dial peer to support out

dtmf-relay [cisco-rtp] [h245- of band signaling of DTMF tones.
signal] [h245-alphanumeric]

Step 5 uBR924(config-dial-peer)# Exit dial-peer configuration mode.

exit

Step 6 To configure incoming calls Specify a unique id-numb er for this incoming
on voice port V2: dial-peer and enter dial-peer configuration
mode.
uBR924(config)# dial-peer
voice id-numb er pots

Step 7 uBR924(config-dial-peer)# Specify the telephone number(s) associated

destination-pattern digits with this voice port.

Step 8 uBR924(config-dial-peer)# Specify that voice port V2 is attached to this

port 1 telephony equipment.

Step 9 uBR924(config-dial-peer)# Optionally configure the dial peer to support out

dtmf-relay [cisco-rtp] [h245- of band signaling of DTMF tones.
signal] [h245-alphanumeric]

Step 10 uBR924(config-dial-peer)# Exit dial-peer configuration mode.

exit

Step 11 Repeat for each possible Specify a unique id-numb er for this outgoing
outgoing destination: dial-peer and enter dial-peer configuration
mode.
uBR924(config)# dial-peer
voice id-numb er voip

Step 12 uBR924(config-dial-peer)# Specify the telephone number(s) associated

destination-pattern digits with this dial-peer.

Step 13 uBR924(config-dial-peer)# Specify the destination IP address or hostname

session target [ for this dial-peer. This could be the IP address
ipv4:ipaddress | or hostname for either an IP telephone or
dns:hostname ] another router or host providing voice services.

Step 14 uBR924(config-dial-peer)# ip (Optional) Specify an IP packet precedence

precedence numb er level (1-5) for packets carrying calls to this dial
peer (1-5, where 5 is the highest precedence
for normal IP flows).

Step 15 uBR924(config-dial-peer)# (Optional) Specify the codec algorithm to be

code [ g711alaw | g711ulaw | used for these calls. The default is g711r8
g729r8 ] (8Kbps compression; A-Law and Mu-Law are
64Kbps compression).

Step 16 uBR924(config-dial-peer)# Optionally configure the dial peer to support out

dtmf-relay [cisco-rtp] [h245- of band signaling of DTMF tones.
signal] [h245-alphanumeric]

Step 17 uBR924(config-dial-peer)# Exit dial-peer configuration mode.

exit

Step 18 uBR924# copy running- Save the configuration to nonvolatile memory

config startup-config so that it will not be lost in the event of a reset,
Building configuration... power cycle, or power outage.

Step 19 uBR924# show startup- Display the configuration file that was just
config created.

Note The ID numbers assigned using the dial-peer voice command must be unique but they are local to the
Cisco uBR924 router. These numbers are used only when configuring each particular dial peer and have no
meaning when dialing numbers or routing calls.

The following example shows a Cisco uBR924 router set up to support bridging and a static H.323 dial map with
the following characteristics:

• Voice port V1 is connected to a telephony device that receives calls for the number 4123.

• Voice port V2 is connected to a telephony device that receives calls for the number 4124.

• Outgoing calls to the numbers 6000—6999 are routed to the dial peer at IP address 10.1.71.65.

• Outgoing calls to the numbers 7000—7999 are routed to the dial peer at IP address 10.1.71.75. These calls
are sent with an IP ToS precedence of "5" and using the G.711 Mu-law codec algorithm.

The commands that set up the H.323v2 dial map are shown in bold:

version 12.1

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

hostname ubr924

clock timezone - 3

ip subnet-zero

no ip routing

voice-port 0

input gain -3

voice-port 1

input gain -3

dial-peer voice 1 pots

destination-pattern 4123

port 0

dial-peer voice 2 pots

destination-pattern 4124

port 1

dial-peer voice 1001 voip

destination-pattern 6...

session target ipv4:10.1.71.65

dtmf-relay cisco-rtp h245-signal h245-alphanumeric

dial-peer voice 1002 voip

destination-pattern 7...

ip precedence 5

codec g711ulaw

session target ipv4:10.1.71.75

dtmf-relay cisco-rtp h245-signal h245-alphanumeric

interface Ethernet0

no ip directed-broadcast

no ip route-cache

bridge-group 59

bridge-group 59 spanning-disabled

interface cable-modem0

ip address dhcp

no ip directed-broadcast

no ip route-cache

cable-modem downstream saved channel 537000000 26

bridge-group 59

bridge-group 59 spanning-disabled

ip classless

no ip http server

no service finger

line con 0

exec-timeout 0 0

transport input none

line vty 0 4

login

end

H.323v2 Static Routing Configuration

When the Cisco uBR924 router is operating in routing mode, the configuration of an H.323v2 static dial map uses
the same commands as those given in the "H.323v2 Static Bridging Configuration" section. The only difference is
that calls can terminate and originate on the Ethernet interface, which is not possible in DOCSIS-bridging mode.

The following sample configuration shows a Cisco uBR924 router set up for a static H.323v2 dial map with the
following characteristics:

• Local dial peer 1 specifies that voice port V1 is connected to a telephone or fax machine with the number
6101.

• Local dial peer 2 specifies that voice port V2 is connected to a telephone or fax machine with the number
6102.

• Remote dial peer 101 specifies that calls to numbers 6200-6299 should be routed to IP address 10.1.71.62.

• Remote dial peers 102 and 103 specify that calls to numbers 6101 and 6102 should be routed to IP address
24.1.61.5, which is the IP address for the Cisco uBR924 router's Ethernet interface. This allows the router to
complete calls between voice ports V1 and V2.

The commands related to the dial map are in bold.

version 12.1

no service pad

service timestamps debug uptime

service timestamps log uptime

hostname ubr924

class-map class-default

match any

clock timezone - 3

ip subnet-zero

voice-port 0

voice-port 1

dial-peer voice 1 pots

destination-pattern 6101

port 0

dial-peer voice 2 pots

destination-pattern 6102

port 1

dial-peer voice 101 voip

destination-pattern 62..

session target ipv4:10.1.71.62

dtmf-relay cisco-rtp

dial-peer voice 102 voip

destination-pattern 6101

session target ipv4:24.1.61.5

dial-peer voice 103 voip

destination-pattern 6102

session target ipv4:24.1.61.5

dtmf-relay cisco-rtp

interface Ethernet0

ip address 24.1.61.1 255.255.255.0

no ip directed-broadcast

no ip mroute-cache

interface cable-modem0

ip address dhcp

no ip directed-broadcast

no ip mroute-cache

cable-modem downstream saved channel 537000000 27

no cable-modem compliant bridge

router rip

version 2

network 10.0.0.0

network 24.0.0.0

no auto-summary

no ip classless

ip route 0.0.0.0 0.0.0.0 10.1.71.1

no ip http server

no service finger

line con 0

exec-timeout 0 0

transport input none

line vty 0 4

login

end

Note The above configuration assumes that the DHCP server assigns an IP address to the cable interface that is
in the class A private network (10.0.0.0).

H.323v2 Dynamic Mapping Configuration

When using a Cisco IOS image that supports voice, the Cisco uBR924 router supports using the Registration,
Admission, and Status (RAS) protocol to allow a remote gatekeeper to translate phone numbers (E.164
addresses) to the IP addresses of specific dial peers. This allows the gatekeeper to maintain a central database
of dial peers, so that this information does not have to be entered into static dial maps on every router that is
acting as a voice gateway.

Note The Cisco uBR924 router can use H.323v2 dynamic mapping in either DOCSIS-bridging mode or routing
mode.

The example shown in this section assumes that Cisco Network Registrar (CNR) version 3.0 or higher is being
used as the DHCP server. CNR assigns the E.164 addresses to local voice ports and uses DHCP to define the
E.164 addresses-to-port assignments.

The gatekeeper can be a Cisco router, such as the Cisco 3620, with a Cisco IOS image that supports the
gatekeeper function. The Cisco uBR924 router acts as the H.323v2 gateway and creates the dial peers, starts
H.323 RAS gateway support, and registers the E.164 addresses with the gatekeeper. The gatekeeper resolves
the remote peers' IP addresses when the router sends a request using RAS.

Note Support for RAS and H.323v2 in Cisco gatekeeper products is found in Cisco IOS Release 12.0(5)T or
higher. Support for multiple classes of service when using Cisco uBR7200 CMTS equipment is found in
Cisco 12.0(4)XI or higher.

If you are not using CNR or Cisco gatekeeper products running Cisco IOS Release 12.0(5)T software, use a static
dial-map as shown in the previous H.323 configurations ("H.323v2 Static Bridging Configuration" and "H.323v2
Static Routing Configuration").

You must do the following to configure the Cisco uBR924 router for dynamic mapping:

• Configure the local dial-peers—This is done in the same way as for a static H.323v2 dial map.

• Configure the remote dial-peers—This is done in the same way as for a static H.323v2 dial map, except that
instead of specifying a target IP address or host name, you specify ras as the target.

• Enable the VoIP gateway function using the gateway global configuration command.

• Configure the cable modem interface to be the gateway interface.

These functions are done using the commands shown in the following table:

Command Purpose

Step 1 To configure incoming Specify a unique id-numb er for this incoming dial-
calls on voice port V1: peer and enter dial-peer configuration mode.

uBR924(config)# dial-
peer voice id-numb er
pots

Step 2 uBR924(config-dial- Specify the telephone number(s) associated with this

peer)# destination- voice port.
pattern digits

Step 3 uBR924(config-dial- Specify that voice port V1 is attached to this telephony

peer)# port 0 equipment.

Step 4 uBR924(config-dial- Optionally configure the dial peer to support out of

peer)# dtmf-relay band signaling of DTMF tones.
[cisco-rtp] [h245-signal]
[h245-alphanumeric]

Step 5 uBR924(config-dial- Exit dial-peer configuration mode.

peer)# exit

Step 6 To configure incoming Specify a unique id-numb er for this incoming dial-
calls on voice port V2: peer and enter dial-peer configuration mode.

uBR924(config)# dial-
peer voice id-numb er
pots

Step 7 uBR924(config-dial- Specify the telephone number(s) associated with this

peer)# destination- voice port.
pattern digits

Step 8 uBR924(config-dial- Specify that voice port V2 is attached to this telephony

peer)# port 1 equipment.

Step 9 uBR924(config-dial- Optionally configure the dial peer to support out of

peer)# dtmf-relay band signaling of DTMF tones.
[cisco-rtp] [h245-signal]
[h245-alphanumeric]

Step 10 uBR924(config-dial- Exit dial-peer configuration mode.

peer)# exit

Step 11 Repeat for each Specify a unique id-numb er for this outgoing dial-
possible outgoing peer and enter dial-peer configuration mode.
destination:

uBR924(config)# dial-
peer voice id-numb er
voip

Step 12 uBR924(config-dial- Specify the telephone number(s) associated with this

peer)# destination- dial-peer.
pattern digits

Step 13 uBR924(config-dial- Specify that RAS will be used to resolve the

peer)# session target destination for the dial-peer.
ras

Step 14 uBR924(config-dial- Optionally configure the dial peer to support out of

peer)# dtmf-relay band signaling of DTMF tones.
[cisco-rtp] [h245-signal]
[h245-alphanumeric]

Step 15 uBR924(config-dial- Exit dial-peer configuration mode.

peer)# exit

Step 16 uBR924(config)# Enable the VoIP gateway on the Cisco uBR924 router.
gateway

Step 17 uBR924(config)# Enter interface configuration mode for the cable

interface cable-modem interface.
0

Step 18 uBR924(config-if)# Enter whatever commands are needed to configure

(enter appropriate cable the cable interface such as IP address, downstream
interface configuration channel, whether DOCSIS-bridging is enabled, and
commands) so forth.

Step 19 uBR924(config-if)# Specify that the cable interface is the H.323 Gateway
h323-gateway voip VoIP interface.
interface

Step 20 uBR924(config-if)# Identify the RAS gatekeeper by specifying its

h323-gateway voip id gatekeeper ID (which must match the ID configured
gatekeeper-id on the gatekeeper), its IP address, and the port
ipaddr IP-address port- number which services gateway requests.
numb er

Step 21 uBR924(config-if)# Specify the H.323 ID for this interface. This ID is any
h323-gateway voip string that uniquely identifies this gateway to the
h323-id interface-id gatekeeper. Typically, this is the gateway's name and
domain (such as "[email protected]").

Step 22 uBR924(config-if)# (Optional) Specify a technology prefix to identify the

h323-gateway voip type of service this gateway can provide. If more than
tech-prefix prefix one service is being provided, give this command for
each separate technology prefix. (The prefix is
defined at the gatekeeper and can up to 11
characters long, with the pound sign (#) as the last
character.)

Step 23 uBR924(config-if)# exit Exit interface configuration mode.

Step 24 uBR924# copy running- Save the configuration to nonvolatile memory so that
config startup-config it will not be lost in the event of a reset, power cycle,
Building configuration... or power outage.

Step 25 uBR924# show startup- Display the configuration file that was just created.
config

Note For additional information on the gateway configuration commands, see the document Configuring H.323
VoIP Gateway for Cisco Access Platforms, available on CCO and the Document CD-ROM.

The following configuration shows a Cisco uBR924 router configured for routing mode and using RAS dynamic
mapping with the following characteristics:

• The router's V1 voice port is connected to a telephone or fax machine with the number 1000, and the V2 voice
port is connected to a telephone or fax machine with the number 1001.

• Four remote dial-peers are configured, with the numbers 1000, 1001, 2000, and 2001. All use the G.711 Mu-
Law CODEC and the RAS protocol is used to resolve their number-address mapping. (The local dial-peer
numbers, 1000 and 1001 are included as remote dial-peers to allow the router to forward calls between the two
local dial-peers, as well as between local and remote dial-peers; the router must be in routing mode to support
this.)

• The cable interface is configured as the gatekeeper interface, using the gatekeeper named gatekeeper3620
at the IP address 10.1.70.50 and at port 1719. The router identifies itself as the gateway named uBR924 with a
tech-prefix of 1#.

The commands related to the dial mapping are in bold.

version 12.1

service config

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

hostname uBR924

clock timezone - 4

ip subnet-zero

ip host-routing

voice-port 0

voice-port 1

dial-peer voice 1 pots

destination-pattern 1000

port 0

dial-peer voice 2 pots

destination-pattern 1001

port 1

dial-peer voice 10 voip

destination-pattern 1001

codec g711ulaw

session target ras

dial-peer voice 20 voip

destination-pattern 1000

codec g711ulaw

session target ras

dial-peer voice 30 voip

destination-pattern 2000

codec g711ulaw

session target ras

dial-peer voice 40 voip

destination-pattern 2001

codec g711ulaw

session target ras

gateway

interface Ethernet0

ip address 24.1.0.1 255.255.0.0

no ip directed-broadcast

no ip mroute-cache

interface cable-modem0

ip address dhcp

no ip directed-broadcast

no ip mroute-cache

no keepalive

cable-modem downstream saved channel 477000000 56

no cable-modem compliant bridge

h323-gateway voip interface

h323-gateway voip id gatekeeper3620 ipaddr 10.1.70.50 1719

h323-gateway voip h323-id uBR924

h323-gateway voip tech-prefix 1#

router rip

version 2

network 10.0.0.0

network 24.0.0.0

ip classless

no ip http server

no service finger

line con 0

transport input none

line vty 0 4

end

Note The above configuration assumes that the DHCP server assigns an IP address to the cable interface that is
in the class A private network (10.0.0.0).

SGCP Configuration
When using Cisco IOS Release 12.0(7)T or higher and a software image that supports voice, the Cisco uBR924
router can use the SGCP protocol for routing voice calls. This transfers the dial mapping to an external call agent,
so that the VoIP gateways do not have to be individually configured with the dial mappings.

Note The Cisco uBR924 router can use SGCP in either DOCSIS-bridging mode or routing mode.

You must do the following to configure the Cisco uBR924 router for a dynamic mapping configuration:

• Enable SGCP operation on the Cisco uBR924 router.

• Specify the SGCP call agent's IP address.

• Configure the local dial-peers to be SCGP applications.

• Optionally enable the sending of SNMP traps for SGCP.

Note No configuration of remote dial-peers is needed when using SGCP.

These functions are done using the commands shown in the following table:

Command Purpose

Step 1 To configure incoming Specify a unique id-numb er for this incoming dial-
calls on voice port V1: peer and enter dial-peer configuration mode.

uBR924(config)# dial-
peer voice id-numb er
pots

Step 2 uBR924(config)# Specify that this dial-peer is handled as an SGCP

application SGCPAPP application.

Step 3 uBR924(config-dial- Specify the telephone number(s) associated with

peer)# destination- this voice port.
pattern digits

Step 4 uBR924(config-dial- Specify that voice port V1 is attached to this

peer)# port 0 telephony equipment.

Step 5 uBR924(config-dial- Exit dial-peer configuration mode.

peer)# exit

Step 6 To configure incoming Specify a unique id-numb er for this incoming dial-
calls on voice port V2: peer and enter dial-peer configuration mode.

uBR924(config)# dial-
peer voice id-numb er
pots

Step 7 uBR924(config)# Specify that this dial-peer is handled as an SGCP

application SGCPAPP application.

Step 8 uBR924(config-dial- Specify the telephone number(s) associated with

peer)# destination- this voice port.
pattern digits

Step 9 uBR924(config-dial- Specify that voice port V2 is attached to this

peer)# port 1 telephony equipment.

Step 10 uBR924(config-dial- Exit dial-peer configuration mode.

peer)# exit

Step 11 ubr924(config)# sgcp Enable SGCP operations on the router.

Step 12 ubr924(config)# sgcp Specify the IP address and optional UDP port
call-agent ip-address [ number for the SGCP call-agent. If no port number
port ] is given, the default of 2427 (the well-known SGCP
port number) is used.

Step 13 uBR924(config)# snmp- (Optional) If SNMP management is used for this

server enable traps router, specify that SGCP and related traps be sent
xgcp to the SNMP manager.

Step 14 uBR924# copy running- Save the configuration to nonvolatile memory so

config startup-config that it will not be lost in the event of a reset, power
Building configuration... cycle, or power outage.

Step 15 uBR924# show startup- Display the configuration file that was just created.
config

The following configuration shows a Cisco uBR924 router configured in DOCSIS-bridging mode that uses SGCP
for the routing of its voice calls. The relevant commands are shown in bold.

version 12.1

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

hostname ubr924

clock timezone - 0 6

ip subnet-zero

no ip routing

ip domain-name cisco.com

ip name-server 4.0.0.32

sgcp

sgcp call-agent 10.186.1.36

xgcp snmp sgcp

voice-port 0

voice-port 1

dial-peer voice 100 pots

application SGCPAPP

destination-pattern 5551212

port 0

dial-peer voice 101 pots

application SGCPAPP

destination-pattern 5551213

port 1

process-max-time 200

interface Ethernet0

no ip directed-broadcast

no ip route-cache

no ip mroute-cache

bridge-group 59

bridge-group 59 spanning-disabled

interface cable-modem0

ip address dhcp

no ip directed-broadcast

no ip route-cache

no ip mroute-cache

cable-modem downstream saved channel 699000000 27

bridge-group 59

bridge-group 59 spanning-disabled

ip classless

no ip http server

no service finger

line con 0

transport input none

line vty 0 4

login

end

MGCP Configuration
When using Cisco IOS Release 12.1(3)T and higher software images that support voice, the Cisco uBR924 router
can use the MGCP protocol for routing voice calls. This transfers the dial mapping to an external call agent or to a
Media Gateway Controller, so that the VoIP gateways do not have to be individually configured with the dial
mappings.

Note The Cisco uBR924 router can use MGCP in either DOCSIS-bridging mode or routing mode.

You must do the following to configure the Cisco uBR924 router for MGCP routing of voice calls:

• Enable MGCP operation on the Cisco uBR924 router.

• Specify the MGCP call agent's IP address.

• Configure the local dial-peers to be MCGP applications.

• Optionally specify the MGCP packages to be supported.

• Optionally change a number of MGCP parameters.

Note No configuration of remote dial-peers is needed when using MGCP.

These functions are done using the commands shown in the following table:

Command Purpose

Step 1 To configure incoming calls Specify a unique id-numb er for this incoming dial-
on voice port V1: peer and enter dial-peer configuration mode.

uBR924(config)# dial-peer
voice id-numb er pots

Step 2 uBR924(config)# Specify that this dial-peer is handled as an MGCP

application MGCPAPP application.

Step 3 uBR924(config-dial-peer)# Specify that voice port V1 is attached to this

port 0 telephony equipment.

Step 4 uBR924(config-dial-peer)# Exit dial-peer configuration mode.

exit

Step 5 To configure incoming calls Specify a unique id-numb er for this incoming dial-
on voice port V2: peer and enter dial-peer configuration mode.

uBR924(config)# dial-peer
voice id-numb er pots

Step 6 uBR924(config)# Specify that this dial-peer is handled as an MGCP

application MGCPAPP application.

Step 7 uBR924(config-dial-peer)# Specify that voice port V2 is attached to this

port 1 telephony equipment.

Step 8 uBR924(config-dial-peer)# Exit dial-peer configuration mode.

exit

Step 9 ubr924(config)# mgcp Enable MGCP operations on the router.

Step 10 ubr924(config)# mgcp call- Specify the IP address and optional UDP port
agent ip-address [ port ] number for the MGCP call-agent. If no port
[ service-type sgcp | mgcp number is given, the default is 2427. The default
] service-type is mgcp, but sgcp can be specified
to ignore RSIP error messages.

Step 11 ubr924(config)# mgcp (Optional) Enables the accurate forwarding of

dtmf-relay { codec | low- touchtone digits during a voice call. Use codec to
bit-rate } mode { cisco | specify the G.711 codec or low-bit-rate to specify
out-of-band } the G.729 codec. Use a mode of cisco to transmit
the tones with the Cisco proprietary method; if the
remote gateway is not a Cisco router, use out-of-
band instead.

Step 12 ubr924(config)# mgcp ip- (Optional) Enable IP Type of Services (TOS) for
tos { high-reliability | high- the voice connections, and specify the value for
throughput | low-cost | the IP precedence bit (the default IP precedence
low-delay | precedence is 3).
value }

Step 13 ubr924(config)# mgcp (Optional) Specify the number of milliseconds to

max-waiting-delay value wait after a restart (default of 3000) before
connecting with the call agent. If used, these
values should be staggered among gateways to
avoid having large numbers of gateways
connecting with the call agent at the same time
after a mass restart.

Step 14 ubr924(config)# mgcp (Optional) Enable the transmission and reception

modem passthru { cisco | of modem and fax data. If the remote gateway is a
ca } Cisco router, specify cisco; otherwise, specify ca
(default) to allow the data to pass-through the
call-agent.

Step 15 ubr924(config)# mgcp (Optional) Specify that the Cisco uBR924 router
package-capability { line- supports a particular package capability. Give this
package | dtmf-package | command multiple times to enable multiple
gm-package | rtp-package packages. Use this command before using the
} mgcp default-package command.

Step 16 ubr924(config)# mgcp (Optional) Specify the default package type for the
default-package { line- media gateway; defaults to line-package.
package | dtmf-package |
gm-package }

Step 17 ubr924(config)# mgcp (Optional) Change the jitter buffer packet size in
playout { adaptive init-value milliseconds for MGCP calls, using either an
min-value max-value | adaptive range or a fixed value. The default is
fixed init-value } adaptive 60 4 200.

Step 18 ubr924(config)# mgcp (Optional) Specify the number of times a call

request retries count request message is transmitted to a call agent
before timing out. The default is 3 times.

Step 19 ubr924(config)# mgcp (Optional) Specify the number of milliseconds to

request timeout timeout wait for a response to a request before
retransmitting or timing out the request. The
default is 500 milliseconds.

Step 20 ubr924(config)# mgcp (Optional) Specify the value (in seconds) used in
restart-delay value Restart in Progress (RSIP) messages to indicate
the delay before the connection is torn down. The
default delay is 0 seconds.

Step 21 ubr924(config)# mgcp vad (Optional) Enable Voice Activity Detection (VAD) to
turn silence suppression on. The default disables
VAD.

Step 22 uBR924# copy running- Save the configuration to nonvolatile memory so

config startup-config that it will not be lost in the event of a reset, power
Building configuration... cycle, or power outage.

Step 23 uBR924# show startup- Display the configuration file that was just
config created.

The following configuration shows a Cisco uBR924 router configured in DOCSIS-bridging mode that uses MGCP
for controlling its voice calls. The relevant commands are shown in bold.

version 12.1

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

hostname ubr924

clock timezone - 0 6

ip subnet-zero

no ip routing

ip domain-name cisco.com

ip name-server 10.0.0.32

mgcp

mgcp call-agent 10.186.1.36

mgcp modem passthru ca

mgcp package-capability dtmf-package

mgcp package-capability line-package

mgcp default-package line-package

xgcp snmp sgcp

voice-port 0

voice-port 1

dial-peer voice 100 pots

application MGCPAPP

port 0

dial-peer voice 101 pots

application MGCPAPP

port 1

process-max-time 200

interface Ethernet0

no ip directed-broadcast

no ip route-cache

no ip mroute-cache

bridge-group 59

bridge-group 59 spanning-disabled

interface cable-modem0

ip address dhcp

no ip directed-broadcast

no ip route-cache

no ip mroute-cache

bridge-group 59

bridge-group 59 spanning-disabled

ip classless

no ip http server

no service finger

line con 0

transport input none

line vty 0 4

login

end

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