Cisco 7965G Administration Guide - Page 20

What Networking Protocols Are Used? Chapter 1 An Overview of the Cisco Unified IP Phone Table 1-1 Supported Networking Protocols on the Cisco Unified IP Phone (continued) Networking Protocol Link Layer Discovery Protocol-Media Endpoint Devices (LLDP-MED) Cisco Peer-to-Peer Distribution Protocol (CPPDP) Real-Time Control Protocol (RTCP) Real-Time Transport Protocol (RTP) Session Initiation Protocol (SIP) Skinny Client Control Protocol (SCCP) Session Description Protocol (SDP) Purpose Usage Notes LLDP-MED is an extension of the LLDP standard developed for voice products. The Cisco Unified IP Phone uses LLDP-MED 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/tech/tk652/tk701/technologie s_white_paper0900aecd804cd46d.shtml CPPDP is a Cisco proprietary protocol used to form a peer-to-peer hierarchy of devices. CPPDP is also used to copy firmware or other files from peer devices to neighboring devices. CPPDP is used by the Peer Firmware Sharing feature. RTCP works with Real-Time Transport Protocol (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 Phone Configuration. For more information, see the "Network Configuration" section on page 4-27. 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. 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. 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 Phones use SCCP for call control. You can configure the Cisco Unified IP Phone to use either SCCP or Session Initiation Protocol (SIP). 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 are supported by all endpoints in the conference. 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 these parameters to be configured on the endpoint itself. Cisco Unified IP Phone 7965G and 7945G Administration Guide for Cisco Unified Communications Manager 6.1 1-6 OL-14641-01