Read the information in the following articles before you deploy your device.

The following table shows the physical and operating environment specifications for the Webex Room Phone.

For more information, see the Webex Room Phone Data Sheet (

Table 1. Physical and Operating Specifications


Value or Range

Operating temperature

32° to 104°F (0° to 40°C)

Operating relative humidity

10% to 90% (non-condensing)

Storage temperature

14° to 140°F (–10° to 60°C)


10.9 inches (278 mm)


10.9 inches (278 mm)


2.4 inches (61.3 mm)


3.98 lb (1.809 kg)


IEEE PoE Class 3 via a PoE injector. The phone is compatible with both IEEE 802.3af and 802.3at switch blades and supports both Cisco Discovery Protocol and Link Layer Discovery Protocol - Power over Ethernet (LLDP-PoE).

Security features

Secure boot


Two HDMI cables ship with your phone. A 9.84 feet (3 meter) cable for HDMI-in and a 26.24 feet (8 meter) cable for HDMI-out.

Distance Requirements

The Ethernet Specification assumes that the maximum cable length between each phone and the switch is 330 feet (100 meters).

Webex Room Phone supports the following codecs:

  • G.711 A-law

  • G.711 mu-law

  • G.722

  • G.729a/G.729ab

  • Opus

The Webex Room Phone supports several industry-standard and Cisco network protocols that are required for voice communication. The following table provides an overview of the network protocols that the phones support.

Table 2. Supported Network Protocols on the Webex Room Phone

Network Protocol


Usage Notes

Cisco Discovery Protocol (CDP)

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

A device can use CDP to advertise its existence to other devices and receive information about other devices in the network.

The 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 the phone into the network and have the phone become operational without the need 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.

We recommend 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, see the documentation for your particular Cisco Unified Communications Manager release.


If you cannot use option 150, use DHCP option 66.

Hypertext Transfer Protocol (HTTP)

HTTP is the standard protocol for transfer of information and movement of documents across the Internet and the web.

Phones use HTTP for XML services, provisioning, upgrade and for troubleshooting purposes.

Hypertext Transfer Protocol Secure (HTTPS)

Hypertext Transfer Protocol Secure (HTTPS) is a combination of the Hypertext Transfer Protocol with the SSL/TLS protocol to provide encryption and secure identification of servers.

Web applications with both HTTP and HTTPS support have two URLs configured. phones that support HTTPS choose the HTTPS URL.

A lock icon is displayed to the user if the connection to the service uses HTTPS.

IEEE 802.1X

The IEEE 802.1X standard defines a client-server-based access control and authentication protocol that restricts unauthorized clients from connection 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 phone implements the IEEE 802.1X standard through support for the following authentication methods: EAP-FAST and EAP-TLS.

Internet Protocol (IP)

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

To communicate with 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 phone with Dynamic Host Configuration Protocol (DHCP). If you are not using DHCP, you must manually assign these properties to each phone locally.

The phones support IPv6 address. For more information, see the documentation for your particular Cisco Unified Communications Manager release.

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 phone supports LLDP on the LAN port.

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

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

The phone supports LLDP-MED on the LAN port to communicate information such as:

  • Voice VLAN configuration

  • Device discovery

  • Power management

  • Inventory management

For more information about LLDP-MED support, see LLDP-MED and Cisco Discovery Protocol here.

Real-Time Transport Protocol (RTP)

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

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 enabled by default.

Session Description Protocol (SDP)

SDP is the portion of the SIP protocol that determines which parameters are available during a connection between two endpoints. Conferences are established by using only the SDP capabilities that all endpoints in the conference support.

SDP capabilities, such as codec types, DTMF detection, and comfort noise, are normally configured on a global basis by Cisco Unified Communications Manager or Media Gateway in operation. Some SIP endpoints may allow configuration of these parameters on the endpoint itself.

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.

Secure Real-Time Transfer protocol (SRTP)

SRTP is an extension of the Real-Time Protocol (RTP) Audio/Video Profile and ensures the integrity of RTP and Real-Time Control Protocol (RTCP) packets providing authentication, integrity, and encryption of media packets between two endpoints.

Phones use SRTP for media encryption.

Transmission Control Protocol (TCP)

TCP is a connection-oriented transport protocol.

Phones use TCP to connect to Cisco Unified Communications Manager and to access XML services.

Transport Layer Security (TLS)

TLS is a standard protocol for securing and authenticating communications.

When security is implemented, phones use the TLS protocol when securely registering with the Cisco Unified Communications Manager. For more information, see the documentation for your particular Cisco Unified Communications Manager release.

Trivial File Transfer Protocol (TFTP)

TFTP allows you to transfer files over the network.

On the 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 Setup menu on the phone.

For more information, see the documentation for your particular Cisco Unified Communications Manager release.

User Datagram Protocol (UDP)

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

UDP is used only for RTP streams. SIP signaling on the phones do not support UDP.

Your device supports the following languages:

  • Dutch

  • English (US)

  • English (UK)

  • French (France)

  • French (Canada)

  • German

  • Italian

  • Portuguese (Brazil)

  • Portuguese (Portugal)

  • Spanish (Spain)

  • Spanish (LATAM)

You can use the Webex Network Test tool to test your network connection. The tool is located at It tests the following network attributes:

  • TCP Connectivity

  • TCP Delay

  • TCP Download speed

  • TCP Upload speed

  • UDP Connectivity

  • UDP Delay

  • UDP Loss Rate