Cisco Unified IP Phone

Phone Overview

The Cisco Unified IP Phone 6901 and 6911 provides voice communication over an Internet Protocol (IP) network. The Cisco Unified IP Phone functions much like a digital business phone, allowing you to place and receive phone calls. The Cisco Unified IP Phones in this guide have the following attributes:

  • Cisco Unified IP Phone 6901 supports basic features such as hold, redial, transfer, and conference.

  • Cisco Unified IP Phone 6911 supports advanced features such as mute, hold, transfer, conference, speed dial, call forward, and more.

A Cisco Unified IP  Phone, like other network devices, must be configured and managed. These phones encode G.711a, G.711mu, G.729a, G.729ab, and iLBC codecs, and decode G.711a, G.711mu, G.729, G.729ab, and iLBC codecs. These phones encode and decode the codecs in similar manner.


Caution

Using a cell, mobile, or GSM phone, or two-way radio in close proximity to a Cisco Unified IP Phone might cause interference. For more information, refer to the manufacturer’s documentation of the interfering device.

For more information, see the Cisco Unified IP Phone 6901 and 6911 User Guide for Cisco Unified Communications Manager (SCCP and SIP).

Cisco Unified IP Phone 6901

The following sections describe the Cisco Unified IP Phone 6901 hardware.

Phone Connections

For your phone to work, it must be connected to the corporate IP telephony network.



1

Slot for Ethernet cable.

4

Network port (10/100 SW) connection. IEEE 802.3af power enabled.

2

Handset connection.

5

DC adaptor port (DC48V).

3

Slot for handset cable.

6

Slot for DC adaptor cable.

Buttons and Hardware



1

Hookswitch

Activates the features (hookflash) on your phone.

2

Hold button

Places a connected call on hold.

3

Redial button

Dials the last dialed number.

4

Line button

Allows you to pick up a second incoming call. The Line button LED indicates the call status.

Allows you to answer a ringing call and swap between two calls on the same line. Also, you can use the line button to create a new call when the phone is idle. The LED associated with the line button lights up to reflect the line status.

Color LEDs indicate the line state:

  • Green, steady—Active call
  • Green, flashing—Held call
  • Amber, Flashing—Incoming call
  • Amber, steady—Call Forward All activated
  • Red, steady—Remote line in use (shared line)
  • Red, flashing—Remote line on hold

5

Volume button

Controls the handset (off hook) and the ringer volume (on hook).

6

Keypad

Allows you to dial phone numbers.

7

Handset with light strip

Lights up to indicate a ringing call (flashing red) or a new voice message (steady red).

Hookswitch Button

The cradle rest of your phone contains the hookswitch button. You can press and quickly release the hookswitch button to activate features (hookflash) on your phone.

Cisco Unified IP Phone 6911

The following sections describe the Cisco Unified IP Phone 6901 hardware.

Phone Connections

For your phone to work, it must be connected to the corporate IP telephony network.



1

4

Network port (10/100 SW) connection. IEEE 802.3af power enabled.

2

5

3

AC power wall plug (optional).

6

Handset connection.

Buttons and Hardware



1

Handset with light strip

Lights up to indicate a ringing call (flashing red) or a new voice message (steady red).

2

Paper label

A paper strip used to enter name and contact numbers.

3

Transfer button

Transfers a call.

4

Conference button

Creates a conference call.

5

Hold button

Places an active call on hold.

6

Line button

Allows you to pick up a second incoming call and to resume a held call. The LED shows call status.

7

Speakerphone button

Selects the speakerphone as the default audio path and initiates a new call, picks up an incoming call, or ends a call. During a call, the button lights green. The speakerphone audio path does not change until you select a new audio path (for example, by picking up the handset).

8

Keypad

Allows you to dial phone numbers.

9

Mute button

Toggles the microphone on or off. When the microphone is muted, the button lights red.

10

Volume button

Controls the handset and speakerphone volume (off hook) and the ringer volume (on hook).

11

Messages button

Auto dials your voice messaging system.

12

Redial button

Dials the last dialed number.

13

Feature button

Depending on the phone setup, the feature button provides you with access to Speed Dial, Call Forward All, Pickup, Group Pickup and Meet Me features. You can configure up to nine items on the feature button. To access these features, press the feature button followed by the number associated with the feature. You must press the feature button and the number within five seconds of each other. The number can only be a single digit number from 1–9.

You can access the following features either off hook or on hook:
  • Call Forward All—Allows you to forward a call.
  • Pickup—Allows you to pickup a call on the third party phone.
  • Group Pickup—Allows you pick up a call within a group.
  • Meet Me—Allows you setup a conference.

14

Handset

Phone handset.

Paper Label

Cisco Unified IP Phone 6911 does not include an LCD display. Cisco provides a paper strip, which can be used to enter name and contact numbers.

Clean the Phone Screen

Procedure

If your phone screen gets dirty, wipe it with a soft, dry cloth.

Caution 

Do not use any liquids or powders on the phone because they can contaminate the phone components and cause failures.

General Phone Information

This section contains information that is common to all the IP Phone models in this guide.

Footstand

If the phone is placed on a table or desk, the footstand can be connected to the back of your phone for a higher or lower viewing angle, depending on your preference.



1

Insert the connectors into the lower slots.

2

Lift the footstand until the connectors snap into the upper slots.

Phone Display Angle

Raise Phone Angle

Procedure

Connect the footstand to the lower slots for a higher viewing angle, as shown in the following figure.



Lower Phone Angle

Procedure

Connect the footstand to the upper slots for a lower viewing angle, as shown in the following figure.



Important Headset Safety Information

High Sound Pressure—Avoid listening to high volume levels for long periods to prevent possible hearing damage.

When you plug in your headset, lower the volume of the headset speaker before you put the headset on. If you remember to lower the volume before you take the headset off, the volume will start lower when you plug in your headset again.

Be aware of your surroundings. When you use your headset, it may block out important external sounds, particularly in emergencies or in noisy environments. Don’t use the headset while driving. Don’t leave your headset or headset cables in an area where people or pets can trip over them. Always supervise children who are near your headset or headset cables.

Network Protocols

Cisco Unified IP Phones support several industry-standard and Cisco network protocols required for voice communication. The following table provides an overview of the network protocols that the Cisco Unified IP Phone 6901 and 6911 support.

Table 1. Supported Network Protocols on the Cisco Unified IP Phone

Network Protocol

Purpose

Usage notes

Cisco Audio Session Tunneling (CAST)

(Cisco Unified IP Phone 6911 only)

The CAST protocol allows IP Phones and associated applications behind the phone to discover and communicate with the remote endpoints without requiring changes to the traditional signaling components like Cisco Unified Communications Manager and gateways. The CAST protocol allows separate hardware devices to synchronize related media and it allows PC applications to augment nonvideo-capable phones to become video enabled by using the PC as the video resource.

-

Cisco Discovery Protocol (CDP)

CDP is a device-discovery protocol that runs on all Cisco-manufactured equipment.

Using CDP, a device advertises its existence to other devices and receives information about other devices in the network.

The Cisco Unified IP Phone uses CDP to communicate information such as auxiliary VLAN ID, per port power management details, and Quality of Service (QoS) configuration information with the Cisco Catalyst switch.

Dynamic Host Configuration Protocol (DHCP)

DHCP dynamically allocates and assigns an IP address to network devices.

DHCP enables you to connect an IP Phone into the network and have the phone become operational without needing to manually assign an IP address or to configure additional network parameters.

By default, the phone has DHCP enabled. If disabled, you must manually configure the IP address, subnet mask, gateway, and a TFTP server on each phone locally.

Cisco recommends that you use DHCP custom option 150. With this method, you configure the TFTP server IP address as the option value. For additional supported DHCP configurations, go to the "Dynamic Host Configuration Protocol" chapter and the "Cisco TFTP" chapter in the Cisco Unified Communications Manager System Guide.

Note 

If you cannot use option 150, you may try using DHCP option 66.

Hypertext Transfer Protocol (HTTP)

HTTP is the standard way of transferring information and moving documents across the Internet and the web.

Cisco Unified IP Phones use HTTP for troubleshooting purposes.

IEEE 802.1X

The IEEE 802.1X standard defines a client-server-based access control and authentication protocol that restricts unauthorized clients from connecting to a LAN through publicly accessible ports.

Until the client is authenticated, 802.1X access control allows only Extensible Authentication Protocol over LAN (EAPOL) traffic through the port to which the client connects. After successful authentication, normal traffic passes through the port.

The Cisco Unified IP Phone implements the IEEE 802.1X standard by providing support for the following authentication methods: EAP-FAST and EAP-TLS.

When 802.1X authentication is enabled on the phone, you should disable the voice VLAN. Refer to the 802.1X Authentication for additional information.

Internet Protocol (IP)

IP is a messaging protocol that addresses and sends packets across the network.

To communicate using IP, network devices must have an assigned IP address, subnet, and gateway.

IP addresses, subnets, and gateways are automatically assigned if you are using the Cisco Unified IP Phone with Dynamic Host Configuration Protocol (DHCP). If you are not using DHCP, you must manually assign these properties to each phone locally.

The Cisco Unified IP Phones support IPv6 address. For more information, see "Internet Protocol Version 6 (IPv6)" in the Cisco Unified Communications Manager Features and Services Guide.

Link Layer Discovery Protocol (LLDP)

(Cisco Unified IP Phone 6911 only)

LLDP is a standardized network discovery protocol (similar to CDP) that is supported on some Cisco and third-party devices.

The Cisco Unified IP Phone 6911 supports LLDP on the switch and PC port.

Link Layer Discovery Protocol-Media Endpoint Devices (LLDP-MED)

LLDP-MED is an extension of the LLDP standard developed for voice products.

The Cisco Unified IP Phone supports LLDP-MED on the SW port to communicate information such as:

  • Voice VLAN configuration
  • Device discovery
  • Power management
  • Inventory management

For more information about LLDP-MED support, see the LLDP-MED and Cisco Discovery Protocol white paper:

http://www.cisco.com/en/US/technologies/tk652/tk701/technologies_white_paper0900aecd804cd46d.html

Real-Time Transport Protocol (RTP)

RTP is a standard protocol for transporting real-time data, such as interactive voice and video, over data networks.

Cisco Unified IP Phones use the RTP protocol to send and receive real-time voice traffic from other phones and gateways.

Real-Time Control Protocol (RTCP)

RTCP works in conjunction with RTP to provide QoS data (such as jitter, latency, and round trip delay) on RTP streams.

By default, the phones have RTCP disabled, but you can enable it on each individual phone using Cisco Unified Communications Manager.

Session Initiation Protocol (SIP)

SIP is the Internet Engineering Task Force (IETF) standard for multimedia conferencing over IP. SIP is an ASCII-based application-layer control protocol (defined in RFC 3261) that can be used to establish, maintain, and terminate calls between two or more endpoints.

Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call.

You can configure the Cisco Unified IP Phone to use either SIP or Skinny Client Control Protocol (SCCP).

Skinny Client Control Protocol (SCCP)

SCCP includes a messaging set that allows communications between call control servers and endpoint clients such as IP Phones. SCCP is proprietary to Cisco Systems.

Cisco Unified IP Phone 6901 and 6911 use SCCP, version 20 for call control.

Transmission Control Protocol (TCP)

TCP is a connection-oriented transport protocol.

Cisco Unified IP Phones use TCP to connect to Cisco Unified Communications Manager.

Transport Layer Security (TLS)

TLS is a standard protocol for securing and authenticating communications.

When security is implemented, Cisco Unified IP Phones use the TLS protocol when securely registering with Cisco Unified Communications Manager.

For more information, see the Cisco Unified Communications Manager Security Guide.

Trivial File Transfer Protocol (TFTP)

TFTP allows you to transfer files over the network.

On the Cisco Unified IP Phone, TFTP enables you to obtain a configuration file specific to the phone type.

TFTP requires a TFTP server in your network, which can be automatically identified from the DHCP server. If you want a phone to use a TFTP server other than the one specified by the DHCP server, you must manually assign the IP address of the TFTP server by using the Network Configuration menu on the phone.

For more information, go to the "Cisco TFTP" chapter in the Cisco Unified Communications Manager System Guide.

User Datagram Protocol (UDP)

UDP is a connectionless messaging protocol for delivery of data packets.

Cisco Unified IP Phones transmit and receive RTP streams, which use UDP.

Cisco Unified IP Phone 6901 and 6911 Supported Features

Cisco Unified IP Phones function much like a digital business phone, allowing you to place and receive phone calls. In addition to traditional telephony features, the Cisco Unified IP Phone includes features that enable you to administer and monitor the phone as a network device.

Feature Overview

Cisco Unified IP Phones provide traditional telephony functionality, Call Forward, Transfer, Redial, Conference, and voice message system access. Cisco Unified IP Phones also provide a variety of other features. For more information on the features, see the Cisco Unified IP Phone 6901 and 6911 User Guide for Cisco Unified Communications Manager (SCCP and SIP).

As with other network devices, you must configure Cisco Unified IP Phones to prepare them to access Cisco Unified Communications Manager and the rest of the IP network. By using DHCP, you have fewer settings to configure on a phone, but if your network requires it, you can manually configure information, such as an IP address, TFTP server, and subnet information.

Finally, because the Cisco Unified IP Phone is a network device, you can obtain detailed status information from the phone directly. This information can assist you with troubleshooting any problems users might encounter when using their IP Phones.

Telephony Feature Administration

You can modify additional settings for the Cisco Unified IP Phone from Cisco Unified Communications Manager. Use Cisco Unified Communications Manager to set up phone registration criteria and calling search spaces, among other tasks.

For more information about Cisco Unified Communications Manager, see the Cisco Unified Communications Manager documentation, including Cisco Unified Communications Manager Administration Guide. You can also use the context-sensitive help available within the application for guidance.

Cisco Unified IP Phone Network Parameters

You configure parameters such as DHCP, TFTP, and IP settings on the phone.

Information for End Users

If you are a system administrator, you are likely the primary source of information for Cisco Unified IP Phone users in your network or company. To ensure that you distribute the most current feature and procedural information, familiarize yourself with Cisco Unified IP Phone documentation on the Cisco Unified IP Phone web site.

In addition to providing documentation, you should inform users about the features enabled on the phones, including those specific to your company or network. You should also give users instructions on how to access and customize the features, if appropriate. You might want to use an internal support web site to keep your users informed.

Cisco Unified IP Phones Security Features

Implementing security in the Cisco Unified Communications Manager system prevents identity theft of the phone and Cisco Unified Communications Manager server, prevents data tampering, and prevents call signaling and media stream tampering.

To alleviate these threats, the Cisco IP telephony network establishes and maintains authenticated and encrypted communication streams between a phone and the server, digitally signs files before they are transferred to a phone, and encrypts media streams and call signaling between Cisco Unified IP Phones.

The Cisco Unified IP Phone 6901 and 6911 use the Phone security profile, which defines whether the device is nonsecure, authenticated, or encrypted. For information on applying the security profile to the phone, see the Cisco Unified Communications Manager Security Guide.

If you configure security-related settings in Cisco Unified Communications Manager, the phone configuration file contains sensitive information. To ensure the privacy of a configuration file, you must configure it for encryption. For detailed information, see the "Configuring Encrypted Phone Configuration Files" chapter in Cisco Unified Communications Manager Security Guide.

The following table shows where you can find additional information about security in this and other documents.

Table 2. Cisco Unified IP Phone and Cisco Unified Communications Manager Security Topics

Topic

Reference

Detailed explanation of security, including set up, configuration, and troubleshooting information for Cisco Unified Communications Manager and Cisco Unified IP Phones

Troubleshooting Guide for Cisco Unified Communications Manager and Cisco Unified Communications Manager Security Guide

Security features supported on the Cisco Unified IP Phone

Supported Security Features

Viewing a security profile name

See Supported Security Features for an overview of the security features supported by the Cisco Unified IP Phone 6901 and 6911. For more information about these features and about Cisco Unified Communications Manager and Cisco Unified IP Phone security, refer to the Cisco Unified Communications Manager Security Guide.

Identifying phone calls for which security is implemented

Authenticated, Encrypted, and Protected Phone Calls

TLS connection

Security and the phone startup process

Phone Startup Process

Security and phone configuration files

Cisco Unified Communications Manager Phone Addition Methods

Disabling access to phone web pages

Disable and Enable Web Page Access

Troubleshooting

Deleting the CTL file from the phone

Cisco Unified IP Phone Reset or Restore

Resetting or restoring the phone

Cisco Unified IP Phone Reset or Restore

802.1X Authentication for Cisco Unified IP Phones

Supported Security Features

The following table provides an overview of the security features that the Cisco Unified IP Phone 6901 and 6911 support. For more information about these features and about Cisco Unified Communications Manager and Cisco Unified IP Phone security, see the Cisco Unified Communications Manager Security Guide.


Note

Most security features are available only if the phone contains a certificate trust list (CTL). For more information about the CTL, see the "Configuring the Cisco CTL Client" chapter in Cisco Unified Communications Manager Security Guide.

Table 3. Overview of Security Features

Feature

Description

Image authentication

Signed binary files (with the extension .zz.sgn) prevent tampering with the firmware image before it is loaded on a phone. Tampering with the image causes a phone to fail the authentication process and reject the new image.

Customer-site certificate installation

Each Cisco Unified IP Phone requires a unique certificate for device authentication. Phones include a manufacturing installed certificate (MIC), but for additional security, you can specify in Cisco Unified Communications Manager Administration that a certificate be installed by using the Certificate Authority Proxy Function (CAPF).

Device authentication

Occurs between the Cisco Unified Communications Manager server and the phone when each entity accepts the certificate of the other entity. Determines whether a secure connection between the phone and a Cisco Unified Communications Manager should occur and, if necessary, creates a secure signaling path between the entities by using TLS protocol. Cisco Unified Communications Manager will not register phones unless they can be authenticated by the Cisco Unified Communications Manager.

File authentication

Validates digitally signed files that the phone downloads. The phone validates the signature to make sure that file tampering did not occur after the file creation. Files that fail authentication are not written to flash memory on the phone. The phone rejects unauthenticated files without further processing.

Signaling Authentication

Uses the TLS protocol to validate that no tampering has occurred to signaling packets during transmission.

Manufacturing installed certificate

Each Cisco Unified IP Phone contains a unique manufacturing installed certificate (MIC), which is used for device authentication. The MIC is a permanent unique proof of identity for the phone, and allows Cisco Unified Communications Manager to authenticate the phone.

Secure SRST reference

After you configure a Cisco Unified Survivable Remote Site Telephony (SRST) reference for security and reset the dependent devices in Cisco Unified Communications Manager, the TFTP server adds the SRST certificate to the phone configuration file and sends the file to the phone. A secure phone uses a TLS connection to interact with the SRST-enabled router.

The configuration file uses one of the following extensions:

  • .cnf.xml
  • .cnf.xml.sgn
  • .cnf.xml.enc.sgn

Media encryption

Uses SRTP to ensure that the media streams between supported phones proves secure and that only the intended phone receives and reads the data. Includes creating a media primary key pair for the phones, delivering the keys to the phones, and securing the delivery of the keys while the keys are in transport.

Signaling encryption

Ensures that all SCCP and SIP signaling messages sent between the phone and the Cisco Unified Communications Manager server are encrypted.

Certificate Authority Proxy Function (CAPF)

Implements parts of the certificate generation procedure that are too processing-intensive for the phone, and interacts with the phone for key generation and certificate installation. The CAPF can be configured to request certificates from customer-specified certificate authorities on behalf of the phone, or it can be configured to generate certificates locally.

Security profiles

Defines whether the phone is nonsecure, authenticated, encrypted, or protected.

Encrypted configuration files

Ensures the privacy of phone configuration files.

Optional disabling of the web server functionality for a phone

Prevents access to a phone web page that displays a variety of operational statistics for the phone.

Phone hardening

The following is an additional security option that you control from Cisco Unified Communications Manager:

  • Disabling access to web pages for a phone

802.1X Authentication

Defines the use of 802.1X authentication to request and gain access to the network.

Voice Quality Metrics

MOS LQK

Objective estimate of the Mean Opinion Score (MOS) for Listening Quality (LQK) that ranks audio quality from 5 (excellent) to 1 (bad). This score is based on audible-concealment events due to a frame loss in the preceding 8 seconds of the voice stream.

Note 

The MOS LQK score can vary based on the type of codec that the Cisco Unified IP Phone uses.

Avg MOS LQK

Average MOS LQK score for the entire voice stream.

Min MOS LQK

Lowest MOS LQK score from the start of the voice stream.

Max MOS LQK

Baseline or highest MOS LQK score from the start of the voice stream.

The following codecs provide the corresponding maximum MOS LQK scores under normal conditions with no frame loss:

  • G.711: 4.5
  • G.728/iLBC: 3.9
  • G729A/AB: 3.7

MOS LQK Version

Version of the Cisco-proprietary algorithm used to calculate the MOS LQK scores.

Security Profiles

All Cisco Unified IP Phones that support Cisco Unified Communications Manager use a security profile, which defines whether the phone is nonsecure, authenticated, or encrypted. For information about configuring the security profile and applying the profile to the phone, see the Cisco Unified Communications Manager Security Guide.

To view the phone security mode, you can view the security profile in Cisco Unified Communications Manager.

Authenticated, Encrypted, and Protected Phone Calls

In an authenticated call, all devices participating in the establishment of the call are trusted devices, and authenticated by Cisco Unified Communications Manager.

In an encrypted call, all devices participating in the establishment of the call are trusted devices, and authenticated by Cisco Unified Communications Manager. In addition, call signaling and media streams are encrypted. An encrypted call offers a high level of security, providing integrity and privacy to the call.

If the call is routed through non-IP call legs, for example, PSTN, the call may be nonsecure even though it is encrypted within the IP network.

In a protected call, a security tone plays at the beginning of a call to indicate that the other connected phone is also receiving and transmitting encrypted audio and video (if video is involved). If your call is connected to a non-protected phone, the security tone does not play.


Note

Connections between two phones support protected calling. Some features, such as conference calling and shared lines, are not available when protected calling is configured. Protected calls are not authenticated.

Identify Protected Calls

A protected call establishes when your phone and the phone on the other end are set up for protected calling. The other phone can be in the same Cisco IP network, or on a network outside the IP network. Protected calls can only be made between two phones. Conference calls and other multiple-line calls do not support protected calls.

A protected call establishes using this process:

  1. A user initiates the call from a protected phone (protected security mode).

  2. A security tone plays if the call connects to another protected phone, indicating that both ends of the conversation are encrypted and protected. If the call connects to a non protected phone, then the secure tone does not play.


Note

Connections between two phones support protected calling. Some features, such as conference calling and shared lines are not available when protected calling is configured.

Call Security Interactions and Restrictions

Cisco Unified Communications Manager checks the phone security status when conferences establish and changes the security indication for the conference or blocks the completion of the call to maintain integrity and security in the system.

The following table provides information about changes to call security levels when using Barge for Cisco Unified IP Phone 6911.

Table 4. Call Security Interactions When Using Barge (Cisco Unified IP Phone 6911 Only)

Initiator’s phone security level

Feature used

Call security level

Results of action

Nonsecure

cBarge

Encrypted call

Call barged and identified as nonsecure call

Secure (encrypted)

cBarge

Authenticated call

Call barged and identified as authenticated call

Secure (authenticated)

cBarge

Encrypted call

Call barged and identified as authenticated call

Nonsecure

cBarge

Authenticated call

Call barged and identified as nonsecure call

The following table provides information about changes to conference security levels depending on the initiator phone security level, the security levels of participants, and the availability of secure conference bridges.

Table 5. Security Restrictions with Conference Calls

Initiator’s phone security level

Feature used

Security level of participants

Results of action

Nonsecure

Conference

Encrypted or authenticated

Nonsecure conference bridge

Nonsecure conference

Secure (encrypted or authenticated)

Conference

At least one member is nonsecure

Nonsecure conference

Secure (encrypted)

Conference

All participants are encrypted

Secure encrypted level conference

Secure (authenticated)

Conference

All participants are encrypted or authenticated

Secure authenticated level conference

Nonsecure

cBarge

All participants are encrypted

Conference changes to nonsecure

Nonsecure

Meet Me

Minimum security level is encrypted

Initiator receives the message Does not meet Security Level, and the call rejected.

Secure (encrypted)

Meet Me

Minimum security level is authenticated

Conference accepts encrypted and authenticated calls

Secure (encrypted)

Meet Me

Minimum security level is nonsecure

Only secure conference bridge available and used

Conference accepts all calls

802.1X Authentication

The following sections describe the 802.1X support on the Cisco Unified IP Phones.

Overview

Cisco Unified IP Phones and Cisco Catalyst switches traditionally use Cisco Discovery Protocol (CDP) to identify each other and determine parameters such as VLAN allocation and inline power requirements. CDP does not identify locally attached workstations. Cisco Unified IP Phones provide an EAPOL pass-through mechanism. This mechanism allows a workstation attached to the Cisco Unified IP Phone to pass EAPOL messages to the 802.1X authenticator at the LAN switch. The pass-through mechanism ensures that the IP phone does not act as the LAN switch to authenticate a data endpoint before accessing the network.

Cisco Unified IP Phones also provide a proxy EAPOL Logoff mechanism. In the event that the locally attached PC disconnects from the IP phone, the LAN switch does not see the physical link fail, because the link between the LAN switch and the IP phone is maintained. To avoid compromising network integrity, the IP phone sends an EAPOL-Logoff message to the switch on behalf of the downstream PC, which triggers the LAN switch to clear the authentication entry for the downstream PC.

Cisco Unified IP Phones also contain an 802.1X supplicant. This supplicant allows network administrators to control the connectivity of IP phones to the LAN switch ports. The current release of the phone 802.1X supplicant uses the EAP-FAST, EAP-TLS, and EAP-MD5 options for network authentication.

Required Network Components

Support for 802.1X authentication on Cisco Unified IP Phones requires several components, including:

  • Cisco Unified IP Phone: The phone acts as the 802.1X supplicant, which initiates the request to access the network.

  • Cisco Secure Access Control Server (ACS) (or other third-party authentication server): The authentication server and the phone must both be configured with a shared secret that authenticates the phone.

  • Cisco Catalyst Switch (or other third-party switch): The switch must support 802.1X, so it can act as the authenticator and pass the messages between the phone and the authentication server. After the exchange completes, the switch grants or denies the phone access to the network.

Best Practices-Requirements and Recommendations

  • Enable 802.1X Authentication: If you want to use the 802.1X standard to authenticate Cisco Unified IP Phones, be sure that you have properly configured the other components before enabling the standard on the phone.

  • Configure PC Port: The 802.1X standard does not take into account the use of VLANs and thus Cisco recommends that only a single device should be authenticated to a specific switch port. However, some switches (including Cisco Catalyst switches) support multi-domain authentication. The switch configuration determines whether you can connect a PC to the PC port of the phone.


    Note

    Only Cisco Unified IP Phone 6911 has a PC port.

    • Enabled: If you are using a switch that supports multi-domain authentication, you can enable the PC port and connect a PC to it. In this case, Cisco Unified IP Phones support proxy EAPOL-Logoff to monitor the authentication exchanges between the switch and the attached PC. For more information about IEEE 802.1X support on the Cisco Catalyst switches, refer to the Cisco Catalyst switch configuration guides at:

      http://www.cisco.com/en/US/products/hw/switches/ps708/tsd_products_support_series_home.html

    • Disabled: If the switch does not support multiple 802.1X-compliant devices on the same port, you should disable the PC Port when 802.1X authentication is enabled. If you do not disable this port and subsequently attempt to attach a PC to it, the switch will deny network access to both the phone and the PC.

  • Configure Voice VLAN: Because the 802.1X standard does not account for VLANs, you should configure this setting based on the switch support.

    • Enabled: If you are using a switch that supports multi-domain authentication, you can continue to use the Voice VLAN.

    • Disabled: If the switch does not support multi-domain authentication, disable the Voice VLAN and consider assigning the port to the native VLAN.

Cisco Unified IP Phone Deployment

When deploying a new IP telephony system, system administrators and network administrators must complete several initial configuration tasks to prepare the network for IP telephony service. For information and a checklist for setting up and configuring a Cisco IP telephony network, see the "System Configuration Overview "chapter in Cisco Unified Communications Manager System Guide.

After you set up the IP telephony system and configure system-wide features in Cisco Unified Communications Manager, you can add IP Phones to the system.

Cisco Unified IP Phones Setup in Cisco Unified Communications Manager

To add phones to the Cisco Unified Communications Manager database, you can use:

  • Autoregistration: Not supported if Cisco Unified Communications Manager operates in mixed mode.

  • Cisco Unified Communications Manager

  • Bulk Administration Tool (BAT)

  • BAT and the Tool for Auto-Registered Phones Support (TAPS)

For general information about configuring phones in Cisco Unified Communications Manager, refer to the following documentation:

  • "Cisco Unified IP Phones", Cisco Unified Communications Manager System Guide

  • "Cisco Unified IP Phone Configuration", Cisco Unified Communications Manager Administration Guide

  • "Autoregistration Configuration", Cisco Unified Communications Manager Administration Guide

The following section provides additional information.

Set Up Cisco Unified IP Phone 6901 and 6911 in Cisco Unified Communications Manager

The following procedure provides an overview of configuration tasks for the Cisco Unified IP Phone 6901 and 6911 in Cisco Unified Communications Manager Administration. The procedure presents a suggested order to guide you through the phone configuration process. Some tasks are optional, depending on your system and user needs. For detailed procedures and information, see the sources in the procedure.

Procedure
Step 1

Gather the following information about the phone:

  • Phone Model

  • MAC address

  • Physical location of the phone

  • Name or user ID of phone user

  • Device pool

  • Partition, calling search space, and location information

  • Associated directory number (DN) to assign to the phone

  • Cisco Unified Communications Manager user to associate with the phone

Provides list of configuration requirements for setting up phones.

For more information, see the "Cisco Unified IP Phones" chapter in the Cisco Unified Communications Manager System Guide and see Telephony Features Available for Cisco Unified IP Phone.

Step 2

Verify that you have sufficient unit licenses for your phone. For more information, see the "License Unit Report"chapter in the Cisco Communications Manager Administration Guide.

Step 3

Add and configure the phone by completing the required fields in the Phone Configuration window. An asterisk (*) next to the field name indicates a required field for example, MAC address and device pool.

The phone with the default settings gets added to the Cisco Unified Communications Manager database. For more information, see the "Cisco Unified IP Phone Configuration" chapter in the Cisco Communications Manager Administration Guide.

For information about Product Specific Configuration fields, see ? Button Help in the Phone Configuration window.

Note 

If you want to add both the phone and user to the Cisco Unified Communications Manager database at the same time, see the "User/Phone Add Configuration" chapter in the Cisco Communications Manager Administration Guide.

Step 4

Add and configure directory numbers (lines) on the phone by completing the required fields in the Directory Number Configuration window. An asterisk (*) next to the field name indicates a required field for example, directory number and presence group.

For more information, see the "Directory Number Configuration" chapter in the Cisco Unified Communications Manager Administration Guide and see Telephony Features Available for Cisco Unified IP Phone.

Step 5

Add user information by configuring required fields. An asterisk (*) next to the field name indicates a required field for example, User ID and last name.

Note 

Assign a password (for User Options web pages) and PIN (for accessing the Network Menu from the Interactive Voice Response [IVR]).

Adds user information to the global directory for Cisco Unified Communications Manager (Unified CM).

For more information, see the "End User Configuration" chapter in the Cisco Unified Communications Manager Administration Guide and see Add Users to Cisco Unified Communications Manager.

Note 

If you want to add both the phone and user to the Cisco Unified Communications Manager database at the same time, see the "User/Phone Add Configurations" chapter in the Cisco Unified Communications Manager Administration Guide.

Step 6

Associate a user to a user group. Assigns users a common list of roles and permissions that apply to all users in a user group. Administrators can manage user groups, roles, and permissions to control the level of access (and, therefore, the level of security) for system users.

Note 

For end users to access Cisco Unified CM User Options, you must add users to the standard Cisco CCM End Users group.

For more information, see the following sections in the Cisco Unified Communications Manager Administration Guide:

  • End User Configuration Settings section in the "End User Configuration" chapter.

  • Adding Users to a User Group section in the "User Group Configuration" chapter.

Step 7

Associate a user with a phone. Provides users with control over their phone for actions such as forwarding calls or adding speed-dial numbers or services.

Note 

Some phones, such as those in conference rooms, do not have an associated user.

For more information, see the Associating Devices to an End User section in the "End User Configuration" chapter in the Cisco Unified Communications Manager Administration Guide.

Cisco Unified IP Phones Installation

After you have added the phones to the Cisco Unified Communications Manager database, you can complete the phone installation. You (or the phone user) can install the phone at the location of the user.


Note

Upgrade the phone with the current firmware image before you install the phone. For information about upgrading, see the Readme file for your phone, located at:

http://tools.cisco.com/support/downloads/go/Redirect.x?mdfid=278875240

For instructions on upgrading the firmware, see the Release Notes, located at:

http://www.cisco.com/en/US/products/ps10326/prod_release_notes_list.html

After the phone connects to the network, the phone startup process begins, and the phone registers with Cisco Unified Communications Manager. To finish installing the phone, configure the network settings on the phone depending on whether you enable or disable DHCP service.

If you used autoregistration, you need to update the specific configuration information for the phone such as associating the phone with a user, changing the button table, or directory number.

Install Cisco Unified IP Phone 6901 and 6911

The following procedure provides an overview of installation tasks for the Cisco Unified IP Phone 6901 and 6911. The procedure presents a suggested order to guide you through the phone installation. Some tasks are optional, depending on your system and user needs. For detailed procedures and information, see the sources in the procedure.

Procedure
Step 1

Choose the power source for the phone:

  • Power over Ethernet (PoE)

  • External power supply

Determines how the phone receives power. For more information, see Cisco Unified IP Phone Power.

Step 2

Assemble the phone, adjust phone placement, and connect the network cable.

Locates and installs the phone in the network. See Install Cisco Unified IP Phone.

See Footstand.

Step 3

Monitor the phone startup process. Associates directory numbers to the phone and verifies that phone is configured properly.

See Phone Startup Verification.

Step 4

If you are configuring the network settings on the phone, you can set up an IP address for the phone by either using DHCP or manually entering an IP address.

  • Using DHCP

    Verify that the phone has DHCP enabled using the IVR. You can set an alternate TFTP server by entering the IP address for the TFTP when prompted by the IVR.

    Note 

    Consult with the network administrator to determine whether you need to assign an alternative TFTP server instead of using the TFTP server assigned by DHCP.

  • Without DHCP

    Verify that the phone has DHCP disabled using the IVR. You must then configure the IP address, subnet mask, TFTP server, and default router locally by using the IVR on the phone.

For more information, see Network Settings and Cisco Unified IP Phone Network Settings Setup.

Step 5

Set up security on the phone.

Provides protection against data tampering threats and identity theft of phones.

For more information, see Cisco Unified IP Phone Security.

Step 6

Make calls with the Cisco Unified IP Phone. Verifies that the phone and features work correctly. For more information, see Cisco Unified IP Phone 6901 and 6911 User Guide for Cisco Unified Communications Manager (SCCP and SIP).

Step 7

Provide information to end users about how to use their phones and how to configure their phone options.

Ensures that users have adequate information to successfully use their Cisco Unified IP Phones.

See Internal Support Web Site

Phone Power Reduction

The Cisco Unified IP Phone 6901 and 6911 supports Cisco EnergyWise (EW) (also known as Power Save Plus). When your network contains an EnergyWise controller, you can configure these phones to sleep (power down) and wake (power up) on a schedule to reduce your power consumption.

You set up each phone to enable or disable the EnergyWise settings. If EnergyWise is enabled, you configure a sleep and wake time, as well as other parameters. These parameters are sent to the phone as part of the phone configuration file.

Terminology Differences

The following table highlights some of the important differences in terminology used in these documents:

  • Cisco Unified IP Phone 6901 and 6911 User Guide for Cisco Unified Communications Manager (SCCP and SIP)

  • Cisco Unified IP Phone 6901 and 6911 Administration Guide for Cisco Unified Communications Manager (SCCP and SIP)

  • Cisco Unified Communications Manager Administration Guide

  • Cisco Unified Communications Manager System Guide

User Guide

Administration and System Guides

Auto Barge

cBarge

Message Indicators

Message Waiting Indicator (MWI) or Message Waiting Lamp

Voice mail system

Voice messaging system