Network Requirements
For the phone to successfully operate as an endpoint in your network, your network must meet the following requirements:
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VoIP Network
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VoIP is configured on your Cisco routers and gateways.
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Cisco Unified Communications Manager is installed in your network and is configured to handle call processing.
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IP network that supports DHCP or manual assignment of IP address, gateway, and subnet mask
Note |
The phone displays the date and time from Cisco Unified Communications Manager. If the user turns off Automatic date and time in the Settings application, the time may become out of sync with the server time. |
Network Protocols
The Cisco Wireless IP Phone 8821 and 8821-EX supports several industry-standard and Cisco network protocols required for voice communication. The following table provides an overview of the network protocols that the phones support.
Network protocol |
Purpose |
Usage notes |
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Bluetooth |
Bluetooth is a wireless personal area network (WPAN) protocol that specifies how devices communicate over short distances. |
The phones support Bluetooth 4.0. |
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Bootstrap Protocol (BootP) |
BootP enables a network device, such as the Cisco IP Phone, to discover certain startup information, such as the IP address. |
None |
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Cisco Audio Session Tunnel (CAST) |
The CAST protocol allows Cisco IP Phones and associated applications to discover and communicate with the remote IP Phones without requiring changes to the traditional signaling components, such as Cisco Unified Communications Manager (CM) and gateways. |
The phones use CAST as an interface between CUVA and Cisco Unified Communications Manager using the Cisco IP Phone as a SIP proxy. |
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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 phones use 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. |
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Cisco Peer-to-Peer Distribution Protocol (CPPDP) |
CPPDP is a Cisco proprietary protocol used to form a peer-to-peer hierarchy of devices. This hierarchy is used to distribute firmware files from peer devices to their neighboring devices. |
CPPDP is used by the Peer Firmware Sharing feature. |
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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 the phone to 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 more information, see the documentation for your particular Cisco Unified Communications Manager release.
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Hypertext Transfer Protocol (HTTP) |
HTTP is the standard way of transferring information and moving documents across the Internet and the web. |
The phones use HTTP for XML services and for troubleshooting purposes. |
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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. |
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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 is connected. After authentication is successful, normal traffic can pass through the port. |
The phones implement the IEEE 802.1X standard by providing support for the following authentication methods:
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IEEE 802.11n/802.11ac |
The IEEE 802.11 standard specifies how devices communication over a wireless local area network (WLAN). |
802.11n operates in the 2.4 GHz and 5 GHz band. 802.11ac operates in the 5 GHz band. |
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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 gateway 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 do not support IPv6. |
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Real-Time Transport Protocol (RTP) |
RTP is a standard protocol for transporting real-time data, such as interactive voice, over data networks. |
The phones use the RTP protocol to send and receive real-time voice traffic from other phones and gateways. |
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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. |
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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. |
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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 addresses the functions of signaling and session management within a packet telephony network. Signaling allows transportation of call information across network boundaries. Session management provides the ability to control the attributes of an end-to-end call. |
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Transmission Control Protocol (TCP) |
TCP is a connection-oriented transport protocol. |
The phones use TCP to connect to Cisco Unified Communications Manager and to access XML services. |
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Transport Layer Security (TLS) |
TLS is a standard protocol for securing and authenticating communications. |
Upon security implementation, the phones use the TLS protocol when securely registering with Cisco Unified Communications Manager. |
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Trivial File Transfer Protocol (TFTP) |
TFTP allows you to transfer files over the network. On the Cisco IP Phone, TFTP enables you to obtain a configuration file specific to the phone type. |
TFTP requires a TFTP server in your network that the DHCP server can automatically identify. If you want a phone to use a TFTP server other than the one that the DHCP server specifies, you must manually assign the IP address of the TFTP server by using the Network Configuration menu on the phone. For more information, see the documentation for your particular Cisco Unified Communications Manager release. |
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User Datagram Protocol (UDP) |
UDP is a connectionless messaging protocol for delivery of data packets. |
UDP is used by the phones for signaling. |
Cisco Wireless IP Phone 882x Deployment Guide
The Cisco Wireless IP Phone 882x Deployment Guide contains useful information about the wireless phone in the Wi-Fi environment. You can find the deployment guide at this location: