Cisco CP-7961G-GE Administration Guide - Page 21
SCCP.Cisco Unified IP Phones do not support the, Cisco Unified IP Phones use the RTP protocol
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Chapter 1 An Overview of the Cisco Unified IP Phone What Networking Protocols are Used? Table 1-2 Supported Networking Protocols on the Cisco Unified IP Phone (continued) Networking Protocol Link Layer Discovery Protocol-Media Endpoint Devices (LLDP-MED) Real-Time Transport Protocol (RTP) Real-Time Control Protocol (RTCP) Session Description Protocol (SDP) Purpose Usage Notes LLDP-MED is an extension of the LLDP The Cisco Unified IP Phone supports LLDP-MED standard developed for voice products. 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/tech/tk652/tk701/tech nologies_white_paper0900aecd804cd46d.shtml RTP is a standard protocol for transporting Cisco Unified IP Phones use the RTP protocol to real-time data, such as interactive voice send and receive real-time voice traffic from other and video, over data networks. phones and gateways. 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. For more information, see Network Configuration Menu, page 4-33. 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. 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).Cisco Unified IP Phones do not support the SIP protocol when the phones are operating in IPv6 address mode. 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 Phones use SCCP for call control. You can configure the Cisco Unified IP Phone to use either SCCP or Session Initiation Protocol (SIP). OL-21011-01 Cisco Unified IP Phone Administration Guide for Cisco Unified Communications Manager 8.0 (SCCP and SIP) 1-7