SIP简明教程.ppt

VoIP Signaling and Call Control Configuring SIP Outline Overview Session Initiation Protocol Components of SIP SIP Messages SIP Addressing Call Setup Models Robust Design Cisco Implementation of SIP Configuring SIP on a Cisco Router Monitoring and Troubleshooting Summary Lesson Self-Check SIP and Associated Standards SIP is a simple extensible protocol. SIP is defined in IETF RFC 2543; RFC 3261. SIP creates, modifies, and terminates multimedia sessions with one or more participants. SIP leverages various IETF standards: RTP, Real-Time Transport Control Protocol (RTCP), HTTP, SDP, DNS, SAP, and Real Time Streaming Protocol (RTSP). SIP performs addressing by E.164, e-mail, or DNS service record. SIP is ASCII text-based for easy implementation and debugging. Functional Components SIP Messages SIP Request Messages INVITE: Indicates that a user or service is being invited to participate in a call session ACK: Confirms that a client has received a final response to an INVITE request BYE: Terminates an existing call; can be sent by either UA CANCEL: Cancels pending searches; does not terminate calls that have been accepted OPTIONS: Queries the capabilities of servers REGISTER: Registers the UA with the registrar server of a domain SIP Response Messages 1xx: Informational response 2xx: Successful response 3xx: Redirection response 4xx: Client error response 5xx: Server error response 6xx: Global failure response Addresses Fully qualified domain names sip:jdoe@ E.164 addresses sip:14085551234@; user=phone Mixed addresses sip:14085551234; password=changeme@ sip:jdoe@ Address Registration Address Resolution Direct Call Setup Call Setup Using a Proxy Server Call Setup Using a Redirect Server Survivability Strategies Cisco Implementation of SIP SIP User Agent SIP Dial Peers Example: show Commands Summary SIP uses IETF protocols, including URL, DNS, and TRIP to define aspects of VoIP and multimedia sessions. The two basic components of SIP are UAs and network servers. SIP use