Cisco Systems OL-8978-02 User Manual

CHAP T E R

55

Configuring a VoIP Network

This chapter describes how to configure a Voice-over-IP (VoIP) network on the Catalyst 6500 series
switches.

Note While this chapter introduces a number of Cisco networking products that are related to VoIP, the

primary focus of the chapter is to provide configuration information for integrating the Catalyst 6500
series products into your VoIP network.

Note For complete syntax and usage information for the commands that are used in this chapter, refer to the

Catalyst 6500 Series Switch Command Reference publication.

This chapter consists of these sections:

• Hardware and Software Requirements, page 55-1

• Understanding How a VoIP Network Works, page 55-2

• Understanding How VLANs Work, page 55-8

• Understanding How CDP and VoIP Work, page 55-10

• Configuring VoIP on a Switch, page 55-10

• Using SmartPorts, page 55-38

Hardware and Software Requirements

The hardware and software requirements for the Catalyst 6500 series switches and Cisco CallManager
are as follows:

• Catalyst 4500 series, 5000 family, and Catalyst 6500 series switches running supervisor engine

software release 6.1(1) or later releases

• Catalyst 4500 series and Catalyst 6500 series switches running supervisor engine software

release 8.2(1) or later releases for IEEE 802.3af compliance

• Cisco CallManager release 3.0 or later releases

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Understanding How a VoIP Network Works

Understanding How a VoIP Network Works

A telephony system built on an IP network instead of the traditional circuit-switched private branch
exchange (PBX) network is called an IP PBX system. (See Figure 55-1.) The system’s components are
described in these sections:

• Cisco IP Phone 7960, page 55-2

• Cisco CallManager, page 55-5

• Access Gateways, page 55-5

• How a Call Is Made, page 55-8

Figure 55-1 IP PBX System

Chapter 55 Configuring a VoIP Network

Cisco CallManager

PSTN or PBX

Voice Gateway 200

Analog stations

(phone, fax, modem)

* Catalyst 4000, 5000, and 6000 10/100 modules

10/100BASE-TX Module *

10/100BASE-TX Module

Catalyst 6500 series switches

IP cloud

Digital Trunk Gateway

(WS-X6608-T1/E1)

Analog Station Gateway

(WS-X6624-FXS)

(WS-X6348-RJ45V)

Analog Trunk Gateway

WS-PWR-PNL

PSTN or PBX

Analog stations

(phone, fax, modem)

IP phone

IP phone

PC

PC

38202

Cisco IP Phone 7960

The Cisco IP Phone 7960 provides the connectivity to the IP PBX system. The IP phone has two RJ-45
jacks for connecting to the external devices: a LAN-to-phone jack and a PC-to-phone jack. The jacks use
either Category 3 or Category 5 unshielded twisted-pair (UTP) cable. The LAN-to-phone jack is used to
connect the phone to the LAN using a crossover cable; a workstation or a PC can be connected to the
PC-to-phone jack using a straight-through cable.

The inline power is designed to work in cables from Category 3, Category 4, Category 5, and later up to
100 meters. The inline power works with IBM Token Ring STP cable of 100 meters when used with a
Token Ring to Fast Ethernet adapter (LanTel Silver Bullet SB-LN/VIP-DATA adapter).

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Chapter 55 Configuring a VoIP Network

The IP phone is Dynamic Host Configuration Protocol (DHCP) capable. Optionally, you can program
the IP phone with a static IP address.

The IP phone can be powered by the following sources:

• External power source—Optional transformer and power cord for connecting to a standard wall

receptacle.

• Ethernet switching modules with the voice daughter card installed—Provides the inline power to the

IP phone.

• WS-PWR-PNL (inline-power patch panel)—Provides the inline power to the IP phone. The inline

patch panel allows the IP phone to connect to existing Catalyst 4500 series, 5000 family, and
6500 series 10/100BASE-TX switching modules.

• WS-PWR-PNL (inline-power patch panel)—Provides the inline power to the IP phone. The inline

patch panel allows the IP phone to connect to existing Catalyst 4500 series, 5000 family, and
6500 series 10/100BASE-TX switching modules.

• WS-X6148-RJ-45 10/100 switching module with either the WS-F6K-VPWR inline-power

field-upgrade module or the WS-F6K-FE48-AF inline-power field-upgrade module—Provides the
inline power to the IP phone.

• WS-X6148-RJ-21 10/100 switching module with either the WS-F6K-VPWR inline-power

field-upgrade module or the WS-F6K-FE48-AF inline-power field-upgrade module—Provides the
inline power to the IP phone.

Understanding How a VoIP Network Works

• WS-X6148X2-RJ-45 10/100 switching module with the WS-F6K-FE96-AF inline-power

field-upgrade module—Provides the inline power to the IP phone.

• WS-X6148X2-RJ-21 10/100 switching module with the WS-F6K-FE96-AF inline-power

field-upgrade module—Provides the inline power to the IP phone.

• WS-6548-GE-TX Gigabit Ethernet switching module with either the WS-F6K-VPWR-GE

inline-power field-upgrade module or the WS-F6K-GE48-AF inline-power field-upgrade
module—Provides the inline power to the IP phone.

• WS-6148-GE-TX Gigabit Ethernet switching module with either the WS-F6K-VPWR-GE

inline-power field-upgrade module or the WS-F6K-GE48-AF inline-power field-upgrade
module—Provides the inline power to the IP phone.

Figure 55-2 shows how to connect the Cisco IP Phone 7960 and PCs to the Catalyst 6500 series switch.

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Understanding How a VoIP Network Works

Figure 55-2 Connecting the Cisco IP Phone 7960 to the Catalyst 6500 Series Switch

Chapter 55 Configuring a VoIP Network

10/100BASE-TX Module

(WS-X6348-RJ45V)

10/100BASE-TX Module *

(WS-X6348-RJ45V)

10/100BASE-TX Module *

(WS-X6348-RJ45V)

10/100BASE-TX Module *

(WS-X6348-RJ45V)

Gigabit Ethernet Module with
Inline power daughter card

(WS-6548-GE-TX + WS-F6K-VPWR)

Catalyst 6500 series switches

*

*

IP phone (Example 1)

PC (Example 2)

IP phone PC (Example 3)

IP phone IP phone

* Or any Catalyst 4000, 5000, and 6000

10/100 module using the inline-power
patch panel (WS-PWR-PNL)

PC (Example 4)

38203

The examples shown in Figure 55-2 are described in detail as follows:

• Example 1: Single Cisco IP Phone 7960

Example 1 shows one IP phone that is connected to the 10/100 port on the Catalyst 6500 series
switch. The PC-to-phone jack on the phone is not used. The phone can be powered through the
10/100 port or wall powered.

• Example 2: Single PC

Example 2 shows one PC that is connected to the 10/100 port on the Catalyst 6500 series switch.
The PC is wall powered.

• Example 3: One Cisco IP Phone 7960 and One PC

55-4

Example 3 shows one IP phone that is connected to the 10/100 port on the Catalyst 6500 series
switch and one PC that is connected to the PC-to-phone jack on the phone. The PC behaves as if it
is connected directly to the 10/100 port on the Catalyst 6500 series switch. The phone can be
powered through the 10/100 port or wall powered. The PC must be wall powered.

• Example 4: Two Cisco IP Phone 7960s and One PC

Example 4 shows two IP phones that are connected to the 10/100 port on the Catalyst 6500 series
switch and one PC that is connected to the PC-to-phone jack on the phone. The PC behaves as if it
is connected directly to the 10/100 port on the Catalyst 6500 series switch. The first phone can be
powered through the 10/100 port or wall powered. The second phone and the PC must be wall
powered.

Note For more information on configuring the Cisco IP phones and third-party vendor phones, refer to the

documentation that shipped with the phone.

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Chapter 55 Configuring a VoIP Network

Cisco CallManager

Cisco CallManager is an open and industry-standard call processing system; its software runs on a
Windows NT server and sets up and tears down the calls between the phones, integrating traditional PBX
functionality with the corporate IP network. Cisco CallManager manages the components of the IP PBX
system, the phones, the access gateways, and the resources for such features as call conferencing and
media mixing. Each Cisco CallManager manages the devices within its zone and exchanges information
with the Cisco CallManager in charge of another zone to make the calls possible across multiple zones.
Cisco CallManager can work with the existing PBX systems to route a call over the Public Switched
Telephone Network (PSTN).

Note For information on configuring Cisco CallManager to work with the IP devices that are described in this

chapter, refer to the Cisco CallManager Administration Guide, the Configuration Notes for Cisco
CallManager, and the Cisco CallManager Remote Serviceability Users Guide publications.

Access Gateways

Understanding How a VoIP Network Works

The access gateways allow the IP PBX system to talk to the existing PSTN or PBX systems. The access
gateways consist of analog station gateways, analog trunk gateways, digital trunk gateways, and a
converged voice gateway.

These sections describe the gateways:

• Analog Station Gateway, page 55-5

• Analog Trunk Gateway, page 55-6

• Digital Trunk Gateway, page 55-6

• Converged Voice Gateway, page 55-7

Analog Station Gateway

The Catalyst 6500 series 24-port Foreign Exchange Station (FXS) analog interface module allows the
plain old telephone service (POTS) phones and fax machines to connect to the IP PBX network. The
analog station gateway behaves like the PSTN side for the POTS equipment. It requires an IP address,
is registered with Cisco CallManager in its domain, and is managed by Cisco CallManager.

To configure the analog station interfaces, see the “Configuring VoIP on a Switch” section on

page 55-10. The module features are listed in Table 55-1.

Table 55-1 24-Port FXS Analog Interface Module Features

Digital Signal Processing Per Port

G.711 and G.729 voice encoding

Silence suppression; voice activity detection

Comfort noise generation

Ringer, software programmable frequency and cadence, based on country

DTMF

Signaling, loop start

1

detection

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Understanding How a VoIP Network Works

Table 55-1 24-Port FXS Analog Interface Module Features (continued)

Digital Signal Processing Per Port

Line echo cancellation (32 ms)

Impedance (600 ohms)

Programmable analog gain, signaling timers

Fax pass-through

2

SPAN

or port mirroring support

FXS Interface Features

Address signaling formats: In-band DTMF

Signaling formats: Loop start

Ringing tone: Programmable

Ringing voltage: Programmable, based on country

Ringing frequency: Programmable, based on country

Distance: 500-ohms maximum loop

1. DTMF = dual tone multifrequency

2. SPAN = Switched Port Analyzer

Chapter 55 Configuring a VoIP Network

Analog Trunk Gateway

The Cisco access analog trunk gateways allow the IP PBX to connect to the PSTN or PBX. The gateway
supports up to eight trunks to the PSTN and appears like a phone to the trunk lines coming from the
PSTN. Using this gateway, the IP PBX places an IP call through the PSTN. Similar to the analog station
gateway, the analog trunk gateway provides line echo cancellation and dual tone multifrequency
(DTMF) tone generation and detection. The analog trunk gateway does not provide the ring voltage as
it is not connected to the POTS end devices such as the POTS phones or fax machines. The analog trunk
gateway requires an IP address, is registered with Cisco CallManager in its domain, and is managed by
Cisco CallManager.

To configure the analog trunk gateways, refer to the documentation that shipped with the gateway.

Digital Trunk Gateway

The Catalyst 6500 series 8-port T1/E1 PSTN interface module can support both digital T1/E1
connectivity to the PSTN or transcoding and conferencing. The module requires an IP address, is
registered with Cisco CallManager in its domain, and is managed by Cisco CallManager.

The module software is downloaded from a TFTP server. Depending upon which software you
download, the ports can serve as the T1/E1 interfaces or the ports support transcoding and conferencing.
The transcoding and conferencing functions are mutually exclusive. For every transcoding port in use,
one less conferencing port is available and vice versa.

To configure the 8-port T1/E1 PSTN interfaces, see the “Configuring VoIP on a Switch” section on

page 55-10. The module features are listed in Table 55-2.

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Chapter 55 Configuring a VoIP Network

Table 55-2 8-Port T1/E1 PSTN Interface Module Features

Digital Signal Processing Per T1/E1 Port

G.711 to G.723 and G.729a transcoding (maximum of 8 x 32 channels of transcoding)

Conference bridging, meet-me, and ad-hoc conference modes (maximum of 8 x 16 channels of
conferencing)

Comfort noise generation

Fax pass-through

Silence suppression, voice activity detection

Line echo cancellation

Common channel signaling

For T1: 23 DS0 channels for voice traffic; 24th channel is used for signaling

For E1: 29 DS0 channels for voice traffic; 16th channel is reserved for signaling

Any channel can be configured for common channel signaling

ISDN Primary Rate Interface signaling: Each interface supports 23 channels for T1 and 30 channels
for E1. The default mode is for the 24th T1 channel or 16th E1 channel to be reserved for signaling.
Both network side and user side operation modes are supported.

T1 binary 8-zero substitution/alternate mark inversion (B8ZS/AMI) line coding, u-law or a-law
coding

E1 HDB3 line coding

T1 line bit rate: 1.544 Mbps

Understanding How a VoIP Network Works

E1 line bit rate: 2.048 Mbps

T1 line code: AMI, B8ZS

E1 line code: HDB3

Framing format: D4 superframe and extended superframe

Link Management

1

FDL

is a link management protocol that is used to help diagnose problems and gather statistics on

T1 lines

1. FDL = Facilities Data Link

Converged Voice Gateway

The Cisco Voice Gateway 200 (VG200) allows you to connect the standard POTS phones (connected
directly to the gateway or anywhere on the PSTN) with Cisco IP or any H.323-compliant telephony
devices. When used with Cisco CallManager, the VG200 functions as a Media Gateway Control Protocol
(MGCP) gateway. The Cisco VG200 provides a 10/100BASE-T Ethernet port for connection to the data
network. The following telephony connections are also available:

• One to four Foreign Exchange Office (FXO) ports for connecting to a central office or PBX

• One to four FXS ports for connecting to POTS telephony devices

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Understanding How VLANs Work

• One or two T1 digital ports for connecting to the following:

–

–

–

These ports can be used to integrate a VoIP network with POTS devices, PBXs, or the PSTN.

To configure the Cisco VG200, refer to the documentation that shipped with the gateway.

How a Call Is Made

An IP phone connects to a LAN either through a hub port or a switch port. The IP phone boots up and
uses DHCP to get its IP address and the IP address of its TFTP file server. The IP phone uses its IP
address to talk to the TFTP server and gets its configuration file. The configuration file includes the IP
address of the phone’s Cisco CallManager(s). The phone then talks with Cisco CallManager and
registers itself. Each time a phone boots up, it might get a different IP address. Cisco CallManager knows
how to associate a consistent user phone number to a particular phone by using the MAC address of the
phone. Cisco CallManager always maintains a table mapping the phone MAC address and phone
number. Each time a phone registers, the table is updated with the new IP address. During the
registration, Cisco CallManager downloads the key pad template and the feature capability for the
phone. It tells the phone which run-time image it should use. The phone then goes to the TFTP server to
get its run-time image. Each phone has a dedicated TCP connection to Cisco CallManager called the
control channel. All control information, such as key pressing, goes from the phone to Cisco
CallManager through this channel. Instructions to generate ring tone, busy tone, and so on comes from
Cisco CallManager to the phone through this channel.

Cisco CallManager stores the IP-address-to-phone-number mapping (and vice versa) in its tables. When
a user wants to call another user, the user keys in the called party’s phone number. Cisco CallManager
translates the phone number to an IP address and generates an IP packet version of the ring tone to the
called IP phone through the TCP connection. When the called IP phone receives the packet, it generates
a ring tone. When the user picks up the phone, Cisco CallManager instructs the called IP phone to start
talking with the calling party and removes itself from the loop. From this point on, the call goes between
the two IP phones through the Real-Time Transport Protocol (RTP) which runs over the User Datagram
Protocol (UDP). Because the voice packets are sensitive to delays, TCP is not suitable for voice
transmission because the timeouts and retries increase the delay between the packets. When any change
occurs during the call due to a feature being pressed on one of the phones, or one of the users hanging
up or pressing the flash button, the information goes to Cisco CallManager through the control channel.

Chapter 55 Configuring a VoIP Network

PSTN using FXO emulation

T1 channel bank using FXS emulation

PBX through a trunk (tie) line using ear and mouth (E&M) emulation

If a call is made to a number outside of the IP PBX network, Cisco CallManager routes the call to an
analog or digital trunk gateway which routes it to the PSTN.

Understanding How VLANs Work

This section describes the native VLANs and the auxiliary VLANs. This section uses the following
terminology:

• Auxiliary VLAN—Separate VLAN for IP phones

• Native VLAN—Traditional VLAN for data

• Auxiliary VLAN ID—VLAN ID of an auxiliary VLAN

• Native VLAN ID—VLAN ID of a native VLAN

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Chapter 55 Configuring a VoIP Network

Note For more information about the VLANs, see Chapter 11, “Configuring VLANs.”

Figure 55-3 shows how to connect a Cisco IP Phone 7960 to a Catalyst 6500 series switch.

Figure 55-3 Switch-to-Phone Connections

Catalyst switch

Understanding How VLANs Work

Cisco IP Phone 7960

Phone
ASIC

Workstation/PC

38204

10/100 module

P1

P2

3-port
switch

P3

Access
port

When the IP phone connects to a 10/100 port on the Catalyst 6500 series switch, the access port
(PC-to-phone jack) of the IP phone can be used to connect a PC.

The packets to and from the PC and to and from the phone share the same physical link to the switch and
the same port of the switch. The various configurations are shown in the “Cisco IP Phone 7960” section

on page 55-2).

Introducing the IP-based phones into the existing switch-based networks raises the following issues:

• The current VLANs might be configured on an IP subnet basis, and additional IP addresses might

not be available to assign the phone to a port so that it belongs to the same subnet as other devices
(PC) that are connected to the same port.

• The data traffic present on the VLAN supporting phones might reduce the quality of the VoIP traffic.

You can resolve these issues by isolating the voice traffic onto a separate VLAN on each of the ports that
are connected to a phone. The switch port that is configured for connecting a phone would have separate
VLANs that are configured for carrying the following:

• Voice traffic to and from the IP phone (auxiliary VLAN)

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• Data traffic to and from the PC that is connected to the switch through the access port of the IP phone

(native VLAN)

Isolating the phones on a separate, auxiliary VLAN increases the quality of the voice traffic and allows
a large number of phones to be added to an existing network where there are not enough IP addresses.
A new VLAN means a new subnet and a new set of IP addresses.

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Understanding How CDP and VoIP Work

Understanding How CDP and VoIP Work

Cisco Discovery Protocol (CDP) was enhanced in software release 8.1(1) to facilitate backward
compatibility with the newer, higher-powered Cisco IP phones. With this enhanced CDP, a Cisco IP
phone can negotiate its power requirements to the switch within the CDP packet. The switch uses this
information to ensure that it does not oversubscribe the available power.

We recommend that you enable CDP on the switch so that the switch can correctly detect and supply
power to the IP phones that are connected to it. CDP is enabled on the Catalyst 6500 series switches by
default; however, you should confirm that CDP is enabled when setting up your VoIP network. For more
information on CDP, see Chapter 31, “Configuring CDP.”

Configuring VoIP on a Switch

This section describes the command-line interface (CLI) commands and the procedures that are used to
configure the Catalyst 6500 series switch for VoIP operation:

• Voice-Related CLI Commands, page 55-10

• Configuring Per-Port Power Management, page 55-11

• Configuring the Auxiliary VLANs on Catalyst LAN Switches, page 55-20

Chapter 55 Configuring a VoIP Network

• Configuring the Access Gateways, page 55-23

• Displaying the Active Call Information, page 55-29

• Configuring QoS in the Cisco IP Phone 7960, page 55-31

• Configuring a Trusted Boundary to Ensure Port Security, page 55-33

Note For information on using automatic voice configuration, see the “Using SmartPorts” section on

page 55-38.

Note You must enable CDP on the Catalyst 6500 series switch port that is connected to the IP phone in order

to communicate the auxiliary VLAN ID, per-port power management details, and quality of service
(QoS) configuration information.

Voice-Related CLI Commands

Table 55-3 lists the CLI commands that are described in the configuration procedures.

Table 55-3 Voice-Related CLI Command Module and Platform Support

CLI Commands Ethernet Module1WS-X6608-T1/E12WS-X6624-FXS

Inline-power related commands

set port inlinepower X

set inlinepower defaultallocation This is a switch-level command and does not affect the

show port inlinepower X

4

individual modules.

3

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Table 55-3 Voice-Related CLI Command Module and Platform Support (continued)

Configuring VoIP on a Switch

CLI Commands Ethernet Module1WS-X6608-T1/E12WS-X6624-FXS

3

show environment power XXX

Voice-related commands

set port auxiliaryvlan X/X

show port auxiliaryvlan X/X

set port voice interface XX

show port voice interface XX

show port voice XXX

show port voice fdl X

show port voice active XXX

QoS commands related to voice

set port qos mod/port cos-ext

X/X

set port qos mod/port trust-ext

show port qos X/X

1. Ethernet Module = Ethernet switching module with voice daughter card.

2. WS-X6608-T1 and WS-X6608-E1 = 8-port T1/E1 ISDN PRI modules.

3. WS-X6624-FXS = 24-port FXS analog station interface module.

4. X = Command supported on Catalyst 6500 series switch only; XX = Command supported on Catalyst 4500 series, 5000 family, and
6500 series switches. All modules that are listed in Table 55-3 are supported only on Catalyst 6500 series switches.

Configuring Per-Port Power Management

This section describes the per-port power management and the CLI commands that are used to configure
power management for IP phones.

Note To determine the exact power requirements for your configuration to ensure that you are within the

system power budget, see the “Generating a System Status Report” section on page 22-16.

Note This section applies to the Ethernet switching modules with the voice daughter card only. For

information on powering the IP phones that are connected to the other Ethernet switching modules, refer
to the Catalyst Family Inline-Power Patch Panel Installation Note publication.

For each IP phone that is connected to an Ethernet switching module with a voice daughter card installed,
the module allocates part of the available system power to power up and run the phone. You can apply
the power on an individual port basis.

Only one IP phone can be powered per port; the phone must be connected directly to the switch port. If
a second phone is daisy chained off the phone that is connected to the switch port, the second phone
cannot be powered by the switch.

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Configuring VoIP on a Switch

This section describes the following topics:

• Using show Commands to Display Module Type and Version Information, page 55-12

• Power Management Modes, page 55-13

• Phone Detection Summary, page 55-16

• Setting the Power Mode of a Port or a Group of Ports, page 55-17

• Setting the Default Power Allocation, page 55-17

• Setting the Inline Power Notification Threshold for a Module, page 55-18

• Displaying the Power Status for Modules and Individual Ports, page 55-18

• Displaying the Switch Power Environment for Modules, page 55-19

Using show Commands to Display Module Type and Version Information

To determine if the module has a voice daughter card installed, enter the show module command and
look at the “Sub” field. For example, in the following display, the 10/100BASE-TX module in slot 3 has
a voice daughter card.

To display the module status and information, perform this task in normal mode:

Chapter 55 Configuring a VoIP Network

Task Command

Display the module status and information. show module [mod]

This example shows a submodule field that provides information about the submodules. The inline
power daughter card that is installed on module 3, as shown in the display, is WS-F6K-SVDB-FE, and
the inline power daughter card that is installed on module 6, as shown in the display, is
WS-F6K-VPWR-GE-TX.

Console> (enable) show module
Mod Slot Ports Module-Type Model Sub Status

--- ---- ----- ------------------------- ------------------- --- -------­1 1 2 1000BaseX Supervisor WS-X6K-SUP2-2GE yes ok
3 3 48 10/100BaseTX Ethernet WS-X6548-RJ-45 yes ok
4 4 48 10/100BaseTX Ethernet WS-X6148-RJ45V no ok
6 6 48 10/100/1000BaseT Ethernet WS-X6148-GE-TX yes ok

Mod Module-Name Serial-Num

--- -------------------- ----------­1 SAD04460M9G
3 SAD0447099V
4 SAD061901FL
6 SAD0706025A

Mod MAC-Address(es) Hw Fw Sw

--- -------------------------------------- ------ ---------- ----------------­1 00-d0-c0-d4-04-4e to 00-d0-c0-d4-04-4f 1.1 6.1(2) 7.7(0.82-Eng)
00-d0-c0-d4-04-4c to 00-d0-c0-d4-04-4d
00-02-4a-30-88-00 to 00-02-4a-30-8b-ff
3 00-02-b9-ff-eb-70 to 00-02-b9-ff-eb-9f 0.203 6.3(1) 8.2(1)
4 00-00-00-00-00-00 to 00-00-00-00-00-2f 1.3 5.4(2) 7.7(0.81)
6 00-40-0b-ff-00-00 to 00-40-0b-ff-00-2f 0.304 7.2(1) 8.2(1)

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Mod Sub-Type Sub-Model Sub-Serial Sub-Hw Sub-Sw

--- ----------------------- ------------------- ----------- ------ -----­1 L3 Switching Engine II WS-F6K-PFC2 SAD044302EA 1.0
3 IEEE InlinePower Module WS-F6K-FE48-AF sasdfasdf 0.1 8.1(0)
6 Inline Power Module WS-F6K-VPWR-GE SAD070700GV 0.201 8.1(0)
Console> (enable)

To display the module and submodule versions, perform this task in normal mode:

Task Command

Display the module and submodule versions. show version [mod]

This example shows how to display the module and submodule versions:

Console> (enable) show version 6
Mod Port Model Serial # Versions

--- ---- ------------------- ----------- -------------------------------------­6 48 WS-X6148-GE-TX SAD0706025A Hw :0.304
Fw :7.2(1)
Sw :8.1(0)
WS-F6K-VPWR-GE SAD070700GV Hw :0.201
Sw :8.1(0)
Console>

Configuring VoIP on a Switch

Power Management Modes

Each port is configured through the CLI, SNMP, or a configuration file to be in one of the following
modes. The CLI command is set port inlinepower mod/port {{auto | static | limit} [wattage] | off}.

• auto—Discovery is enabled and the supervisor engine directs the switching module to power up the

port only if the switching module discovers the phone. You can specify the maximum wattage that
is allowed on the port. If you do not specify a wattage, then the switch will deliver no more than the
hardware-supported maximum value.

• static—Discovery is enabled and the supervisor engine directs the switching module to power up

the port to the wattage that you specify only if the switching module discovers the phone. You can
specify the maximum wattage that is allowed on the port. If you do not specify a wattage, then the
switch allows the hardware-supported maximum value. The maximum wattage, whether determined
by the switch or specified by you, is preallocated to the port. If the switch does not have enough
power for the allocation, the command will fail.

• off—Discovery is disabled which prevents the port from providing power to an external device. If the

external device is wall-powered and the inline power is off, the port should still link up, join the bridge
group, and go to the STP forwarding state.

• limit—Discovery is enabled. This mode provides you with the option to limit the power allocated for an

external device. If the wattage value that you specify with the limit keyword is less than the power
determined through IEEE classification, instead of denying power, the minimum of these two values is
allocated. If the device consumes more than the configured value, the port is shut down and an
appropriate syslog message is displayed. The limit keyword is not supported on all modules. To check if
the limit keyword is supported on a module, enter the show environment power mod command. If the
output of the command indicates support for per-port power monitoring, the mode is supported.

• max-wattage—(Optional) The maximum power allowed on the port in either auto or static mode; valid

values are from 4000 to 15400 milliwatts.

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Configuring VoIP on a Switch

Each port also has a status that is defined as one of the following:

• on—Power is supplied by the port.

• off—Power is not supplied by the port.

• Power-deny—The supervisor engine does not have enough power to allocate to the port, or the

• err-disable—The port is unable to provide the power to the connected device that is configured in

• faulty—The port failed the diagnostics tests.

These sections provide the information on the IP phone power requirements and management:

• Power Requirements, page 55-14

• Available Power, page 55-15

• Wall-Powered Phones, page 55-15

• Powering Off the Phone, page 55-15

• Phone Removal, page 55-15

• High-Availability Support, page 55-16

Chapter 55 Configuring a VoIP Network

power that is configured for the port is less than the power that is required by the port; the power is
not being supplied by the port.

Static mode.

Power Requirements

The IP phones may have different power requirements. Table 55-4 lists the power requirements for the
different classes of IP phones. The supervisor engine initially calculates the power allocation for each
port based on the per-port configuration, classification (IEEE only), and default power. When the correct
amount of power is determined from the CDP messaging with the Cisco IP Phone, the supervisor engine
reduces or increases the allocated power for any ports that are set to Auto mode. The allocated power is
not adjusted for ports that are set to Static mode.

For example, the default allocated power is 7 W for a Cisco IP Phone requiring 6.3 W. The supervisor
engine allocates 7 W for the Cisco IP Phone and powers it up. Once the Cisco IP Phone is operational,
it sends a CDP message with the actual power requirement to the supervisor engine. The supervisor
engine then decreases the allocated power to the required amount if the port is set to Auto mode. If the
port is set to Static mode, the supervisor engine allocates the wattage that you specified. If the port is set
to off, the supervisor engine does not allot any power to the port.

Table 55-4 Power Requirements for IP Phones

Phone Class Required Power (W)

Cisco 6.3

Cisco + IEEE 7

Cisco High Power 15.4

Class 0 IEEE 15.4

Class 1 IEEE 4

Class 2 IEEE 7.0

Class 3 15.4

Class 4 Refer to Class 0 Reserved

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Chapter 55 Configuring a VoIP Network

Available Power

Table 55-5 lists the available power that can be supplied for each port for the voice daughter cards.

Table 55-5 Efficiency of Voice Daughter Cards

Daughter Card

WS-F6K-PWR 6.3 100%

WS-F6K-VPWR-GE 6.3 89%

WS-F6K-GE48-AF 15 89%

WS-F6K-FE48-AF 15 89%

WS-F6K-FE96-AF 15 89%

For example, if the powered device requires 6.3 W, then the allotted power for that port using a daughter
card with 89 percent efficiency must be 6.3/(0.89) = 7 W. If you are using a voice daughter card with
100 percent efficiency, then the allotted power is 6.3 W.

Configuring VoIP on a Switch

Maximum Power
Per Port (W) Efficiency

Wall-Powered Phones

Powering Off the Phone

Phone Removal

When a wall-powered phone is present on a switching module port, the switching module cannot detect
its presence. The supervisor engine discovers the phone through CDP messaging with the port. If the
phone supports the inline power (the supervisor engine determines this through CDP), and the mode is
set to Auto, Static, or Off, the supervisor engine does not attempt to power on the port. If a power outage
occurs, and the mode is set to Auto, the phone loses power, but the switching module discovers the phone
and informs the supervisor engine, which then applies the inline power to the phone. If a power outage
occurs, and the mode is set to Static, the phone loses power, but the switching module discovers the
phone and applies the preallocated inline power to the phone.

The supervisor engine can turn off power to a specific port by sending a message to the switching
module. The power for a port in Auto mode is then added back to the available system power. The power
for the ports in Static mode is not added back to the available system power. This situation occurs only
when you power off the phone through the CLI or SNMP.

The switching module informs the supervisor engine if a powered phone is removed using a link-down
message. The supervisor engine then adds the allocated power for that port back to the available system
power.

In addition, the switching module informs the supervisor engine if an unpowered phone is removed.

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Caution When a phone cable is plugged into a port and the power is turned on, the supervisor engine has a

4-second timeout waiting for the link to go up on the line. During those 4 seconds, if the phone cable is
unplugged and a network device is plugged in, the device could be damaged. We recommend that you
wait at least 10 seconds between unplugging a device and plugging in a new device.

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High-Availability Support

To support high availability during a failover from the active supervisor engine to the standby supervisor
engine, the per-port power management and phone status information is synchronized between the active
and standby supervisor engines.

The information to be synchronized (on a per-port basis) is the presence of a phone, the phone power
status (on, off, denied, or faulty), allocated power, device class, device type, device maximum power,
and device discovery. The active supervisor engine sends this information to the standby supervisor
engine, and the standby supervisor engine updates its internal data structures. When a switchover occurs,
the standby supervisor engine allocates the power to the modules and ports from the available power,
one module at a time. Once the power for each module has been allocated, the supervisor engine
allocates the power to the phones, beginning with the lowest slot number, until all inline powered ports
have been either powered on, off, or denied.

Phone Detection Summary

Figure 55-4 shows how the system detects a phone that is connected to a Catalyst 6500 series switch port.

Figure 55-4 Power Detection Summary

Chapter 55 Configuring a VoIP Network

Catalyst Switch

10/100 module

10/100 module

10/100 module

10/100 module

Network

device

Cisco phone

Cisco phone
or third party phone.

Wall-power

Third party phone
without CDP.

Switching module
discovers the phone.

Supervisor engine discovers
the phone through CDP and/or
IEEE.

Switching module will not discover
the phone because CDP is not
supported. However, the supervisor
engine detects the phone and powers
it up.

Phone is inserted but has not booted,
then phone is removed. A network
device is plugged in. Inline power
might damage the network device.

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Network

device

Cisco phone
or third party phone
with CDP.

Wall-power

Supervisor engine discovers the
phone through CDP and/or IEEE.

38205

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Setting the Power Mode of a Port or a Group of Ports

To set the power mode of a port or a group of ports, perform this task in normal mode:

Task Command

Set the power mode of a port or a group of ports. set port inlinepower mod/port {[auto | static]

Note If you configure the max-wattage values that are multiples of 500 on a Catalyst 6500 series switch with

the set port inlinepower mod/port static | auto max-wattage command, the power that is drawn from
the global allocation is possibly slightly smaller than the power that is reported in the Total PWR
Allocated to Module field of the show environment power command. This discrepancy is due to the
internal conversion of units from Watts to cAmps and back to Watts. The difference between the total
allocated power and the total power that is drawn from the system is no more than +/- 0.42 W.

This example shows how to set the power mode of a port or group of ports:

Console> (enable) set port inlinepower 2/5 off
Inline power for port 2/5 set to off.
Console> (enable) set port inlinepower 2/3-9 auto 800
Inline power for ports 2/3-9 set to auto and max-wattage to 800.
Console> (enable)

Configuring VoIP on a Switch

[max-wattage] | off}

Setting the Default Power Allocation

The set inlinepower defaultallocation command is global and only affects Cisco IP phones. The inline
power threshold notification generates a syslog message when the inline power usage exceeds the specified
threshold. To set the default power allocation, perform this task in privileged mode (the default allocation
value is 15400 milliwatts):

Caution The set inlinepower defaultallocation command can be harmful when there is not enough power in the

system to bring up all connected inline power devices. If you set a small value for the power allocation,
all connected inline power devices initially will be powered up. However, after receiving CDP messages,
the system will learn that devices are consuming more power and deny power to some of the ports.
Setting a small value might also result in the overdrawing of power for some time with unanticipated
results, such as hardware failures and unexpected resets.

Note 7000 milliwatts is the maximum power supported for these modules: WS-X6348-RJ21V,

WS-X6348-RJ-45V, WS-X6148-RJ-45V, and WS-X6148-RJ21V.

Task Command

Set the default power allocation. set inlinepower defaultallocation value

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