The Cisco Unified SIP Phone 3905 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. In addition, the Cisco Unified SIP Phone 3905 supports features such as mute, hold, transfer, conference, call forward, and more.
A Cisco Unified IP Phone, like other network devices, must be configured and managed. These phones encode G.711a, G.711µ, G.729a, and G.729ab, and decode G.711a, G.711µ, G.729, G.729a, and G.729ab.
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.
Shows information about your phone such as directory number, active call, and phone menu listings.
Indicates an incoming call (flashing red) or new voice message (steady red).
Navigation bar and Select/Feature button
The Navigation bar allows you to scroll through menus and highlight items. The Select button (in the middle of the Navigation bar) allows you to select a highlighted item.
When the phone is off-hook, the Select button functions as the Feature button. You can access these features:
Call Forward All: Allows you to forward a call.
Voice Mail: Allows you access voice mails.
Call Pickup: Allows you to answer a call that is ringing on a co-worker's phone.
Group Call Pickup: Allows you to answer a call that is ringing in another call group.
Opens or closes the Applications menu. Use it to access call history, user preferences, phone settings, and phone model information.
Transfers a call.
Places an active call on hold or resumes a held call.
Allows you to dial phone numbers.
Selects the speakerphone as the default audio path and initiates a new call, picks up an incoming call, or ends a call. The speakerphone audio path does not change until a new default audio path is selected (for example, by picking up the handset).
Controls the handset and speakerphone volume (off-hook) and the ringer volume (on hook).
Toggles the microphone on or off.
Dials the last dialed number.
Returns to the previous screen or menu.
Displays date and time. Also displays line information such as voicemail and missed calls. When using the Applications menu, displays phone menu listings.
Line details and other phone information
Displays the directory number. During a call, also displays details for the active line. If not on a call, also displays line text label and other information such as placed calls and phone menu listings.
Use the following figure to connect the phone to the network.
DC adapter port (DC 4.2V).
Network port (10/100 SW) connection. IEEE 802.3af power enabled.
AC-to-DC power supply (optional).
Access port (10/100 PC) connection.
AC power wall connection.
The Cisco Unified IP Phone 3905 has a foldable footstand. When the footstand is unfolded, it gives the phone an elevated viewing angle.
Adjust Handset Rest
If your phone is wall-mounted, you may need to adjust the handset rest to ensure that the receiver does not slip out of the cradle.
Remove the handset from the cradle and pull the plastic tab from the handset rest.
Rotate the tab 180 degrees.
Hold the tab between two fingers, with the corner notches facing you.
Line up the tab with the slot in the cradle and press the tab evenly into the slot. An extension protrudes from the top of the rotated tab.
Return the handset to the handset rest.
Supported Networking Protocols
Cisco Unified IP Phones support several industry-standard and Cisco networking protocols required for voice communication. The following table provides an overview of the networking protocols that the Cisco Unified SIP Phone 3905 support.
Table 1 Supported Networking Protocols on the Cisco Unified IP Phone
Cisco Discovery Protocol (CDP)
CDP is a device-discovery protocol that runs on all Cisco-manufactured equipment.
Using CDP, a device can advertise its existence to other devices and receive 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 your needing to manually assign an IP address or to configure additional network parameters.
DHCP is enabled by default. 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.
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.
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 is connected. After authentication is successful, normal traffic can pass through the port.
The Cisco Unified IP Phone implements the IEEE 802.1X standard by providing support for the MD5 authentication method.
When 802.1X authentication is enabled on the phone, you should disable the voice VLAN. See the Security Configuration Menu 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 identifications 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.
Link Layer Discovery Protocol (LLDP)
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 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
For more information about LLDP-MED support, see the LLDP-MED and "Cisco Discovery Protocol" white paper:
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.
RTCP is disabled by default, but you can enable it on a per phone basis by 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.
Transmission Control Protocol (TCP)
TCP is a connection-oriented transport protocol.
Cisco Unified IP Phones use TCP to connect to Cisco Unified Communications Manager.
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 utilize UDP.
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.
Cisco Unified IP Phones provide traditional telephony functionality, such as call forwarding and transferring, redialing, conference calling, and voice messaging system access. Cisco Unified IP phones also provide a variety of other features.
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 an IP address, TFTP server, subnet information, and so on.
Finally, because the Cisco Unified IP Phone is a network device, you can obtain detailed status information from it directly. This information can assist you with troubleshooting any problems users might encounter when using their IP phones.
You can modify additional settings for the Cisco Unified IP Phone from Cisco Unified Communications Manager Administration. Use Cisco Unified Communications Manager Administration to set up phone registration criteria and calling search spaces, among other tasks. See the "Telephony Features" section in this document and the Cisco Unified Communications Manager documentation for additional information.
For more information about Cisco Unified Communications Manager Administration, 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.
You can access Cisco Unified Communications Manager documentation at this location:
You configure parameters such as DHCP, TFTP, and IP settings on the phone itself. For more information about configuring settings and viewing statistics from the phone, see Cisco Unified IP Phone Settings.
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 3905 web site:
From this site, you can view various user documentation.
In addition to providing documentation, it is important to inform users of available Cisco Unified IP Phone features - including those specific to your company or network - and of how to access and customize those features, if appropriate.
For a summary of some of the key information that phone users need their system administrators to provide, see Internal Support Web Site
Cisco Unified IP Phone Security Features
The following table shows where you can find information about security in this and other documents.
Table 2 Cisco Unified IP Phone and Cisco Unified Communications Manager Security Topics
Detailed explanation of security, including set up, configuration, and troubleshooting information for Cisco Unified Communications Manager and Cisco Unified IP Phones
See the Troubleshooting Guide for Cisco Unified Communications Manager
Security features supported on the Cisco Unified IP Phone
The following table provides an overview of the security features that the Cisco Unified SIP Phone 3905 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.
Table 3 Overview of Security Features
Optional disabling of the web server functionality for a phone
You can prevent access to a phone web page, which displays a variety of operational statistics for the phone.
The Cisco Unified IP Phone can use 802.1X authentication to request and gain access to the network. See the 802.1X authentication for more information.
Voice Quality Metrics
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.
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:
MOS LQK Version
Version of the Cisco-proprietary algorithm used to calculate the MOS LQK scores.
The Cisco Unified IP Phones support 802.1X authentication.
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 it on the phone.
Configure PC Port: The 802.1X standard does not take into account the use of VLANs and thus 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 phone’s PC port.
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 Phones 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, go to the "System Configuration Overview" chapter in Cisco Unified Communications Manager System Guide.
After you have set up the IP telephony system and configured system-wide features in Cisco Unified Communications Manager, you can add IP phones to the system.
Set up Cisco Unified SIP Phone 3905 in Cisco Unified Communications Manager
The following list provides an overview and checklist of configuration tasks for the Cisco Unified SIP Phone 3905 in Cisco Unified Communications Manager Administration. The list 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, refer to the sources in the list.
Gather the following information about the phone:
Physical location of the phone
Name or user ID of phone user
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
The information provides a list of configuration requirements for setting up phones and identifies preliminary configuration that you need to perform before configuring individual phones.
For more information, see the Cisco Unified IP Phones chapter in the Cisco Unified Communications Manager System Guide and the Available telephony features.
Verify that you have sufficient unit licenses for your phone.
For more information, go to the License Unit Report chapter in the Cisco Unified Communications Manager Administration Guide.
Add and configure the phone by completing the required fields in the Phone Configuration window. Required fields are indicated by an asterisk (*) next to the field name; for example, MAC address and device pool.
The device with its default settings gets added to the Cisco Unified Communications Manager database.
For more information, go to the Cisco Unified IP Phone Configuration chapter in the Cisco Unified Communications Manager Administration Guide.
For information about Product Specific Configuration fields, refer to ? Button Help in the Phone Configuration window.
If you want to add both the phone and user to the Cisco Unified Communications Manager database at the same time, go to the User/Phone Add Configuration chapter in the Cisco Unified Communications Manager Administration Guide
Add and configure directory numbers (line) on the phone by completing the required fields in the Directory Number Configuration window. Required fields are indicated by an asterisk (*) next to the field name; for example, directory number and presence group.
For more information, go to the Directory Number Configuration chapter in the Cisco Unified Communications Manager Administration Guide and see Available telephony features.
Add user information by configuring required fields. Required fields are indicated by an asterisk (*); for example, User ID and last name.
Assign a password (for User Options web pages) and PIN.
Adds user information to the global directory for Cisco Unified Communications Manager.
If you want to add both the phone and user to the Cisco Unified Communications Manager database at the same time, go to the User/Phone Add Configurations chapter in the Cisco Unified Communications Manager Administration Guide.
Associate a user to a user group. This step 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.
In order for end users to access Cisco Unified CM User Options, you must add users to the standard Cisco CCM End Users group.
Refer to 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.
(Optional)Associate a user with a phone.
Provides users with control over their phone such a forwarding calls or services.
Some phones, such as those in conference rooms, do not have an associated user.
For more information, go to 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 users) can install the phone at the location of the user.
Upgrade the phone with the current firmware image before you install the phone. For information about upgrading, refer to the Readme file for your phone, located at:
After the phone is connected 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 auto-registration, 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.
The following list provides an overview and checklist of installation tasks for the Cisco Unified SIP Phone 3905. The list 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, refer to the sources in the list.
Make calls with the Cisco Unified IP Phone. This step verifies that the phone and features work correctly.
For more information, see Cisco Unified SIP Phone 3905 User Guide for Cisco Unified Communications Manager 9.0 (SIP).
Provide information to end users about how to use their phones and how to configure their phone options. This step ensures that users have adequate information to successfully use their Cisco Unified IP Phones.