SIP is rather a component that can be used with other IETF protocols to
build a complete multimedia architecture. Typically, these
architectures will include protocols such as the Real-time Transport
Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and
providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC
2326 [29]) for controlling delivery of streaming media, the Media
Gateway Control Protocol (MEGACO) (RFC 3015 [30]) for controlling
gateways to the Public Switched Telephone Network (PSTN), and the
Session Description Protocol (SDP) (RFC 2327 [1]) for describing
multimedia sessions. Therefore, SIP should be used in conjunction
with other protocols in order to provide complete services to the
users. However, the basic functionality and operation of SIP does
not depend on any of these protocols.
Max-Forwards serves to limit the number of hops a request can make on
the way to its destination. It consists of an integer that is
decremented by one at each hop.
Content-Type contains a description of the message body (not shown).
Content-Length contains an octet (byte) count of the message body.
The complete set of SIP header fields is defined in Section 20.
本文介绍了SIP(会话发起协议)作为构建完整多媒体架构的一个组件,并与其他IETF协议配合使用,如RTP用于实时数据传输及QoS反馈,RTSP用于流媒体控制等。此外还介绍了SIP消息头部字段,包括限制请求跳数的Max-Forwards字段,以及描述消息体类型的Content-Type字段。
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