Real Time Streaming Protocol

RTSP协议是一种用于娱乐和通信系统的网络控制协议,专为控制流媒体服务器而设计。它允许客户端通过VCR-like命令如播放和暂停来实现对媒体文件的实时控制。尽管与HTTP类似,RTSP定义了控制序列以控制多媒体播放,但它具有状态,并使用TCP进行端到端连接。RTSP协议包括基本请求,如OPTIONS、DESCRIBE、SETUP、PLAY、PAUSE、RECORD和TEARDOWN,用于建立和控制媒体会话。

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 The Real Time Streaming Protocol ( RTSP) is a network control protocol designed for use in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between end points. Clients of media servers issue VCR-like commands, such as play and pause, to facilitate real-time control of playback of media files from the server.

The transmission of streaming data itself is not a task of the RTSP protocol. Most RTSP servers use the Real-time Transport Protocol (RTP) in conjunction with Real-time Control Protocol (RTCP) for media stream delivery, however some vendors implement proprietary transport protocols. The RTSP server from RealNetworks, for example, also features RealNetworks' proprietary Real Data Transport (RDT).

RTSP was developed by the Multiparty Multimedia Session Control Working Group (MMUSIC WG) of the Internet Engineering Task Force (IETF) and published as RFC 2326 in 1998.[1]

RTSP using RTP and RTCP allows for the implementation of rate adaption.[further explanation needed]

Protocol directives

While similar in some ways to HTTP, RTSP defines control sequences useful in controlling multimedia playback. While HTTP is stateless, RTSP has state; an identifier is used when needed to track concurrent sessions. Like HTTP, RTSP uses TCP to maintain an end-to-end connection and, while most RTSP control messages are sent by the client to the server, some commands travel in the other direction (i.e. from server to client).

Presented here are the basic RTSP requests. Some typical HTTP requests, like the OPTIONS request, are also available. The default transport layer port number is 554.

OPTIONS
An OPTIONS request returns the request types the server will accept.
DESCRIBE
A DESCRIBE request includes an RTSP URL (rtsp://...), and the type of reply data that can be handled. The default port for the RTSP protocol is 554 for both UDP (deprecated and very rarely used) and TCP transports. This reply includes the presentation description, typically in Session Description Protocol (SDP) format. Among other things, the presentation description lists the media streams controlled with the aggregate URL. In the typical case, there is one media stream each for audio and video.
C->S: DESCRIBE rtsp://example.com/media.mp4 RTSP/1.0
      CSeq: 1

S->C: RTSP/1.0 200 OK
      CSeq: 1
      Content-Base: rtsp://example.com/media.mp4
      Content-Type: application/sdp
      m=video 0 RTP/AVP 96
      a=control:streamid=0
      a=range:npt=0-7.741000
      a=length:npt=7.741000
      a=rtpmap:96 MP4V-ES/5544
      a=mimetype:string;"video/MP4V-ES"
      a=AvgBitRate:integer;304018
      a=StreamName:string;"hinted video track"
      m=audio 0 RTP/AVP 97
      a=control:streamid=1
      a=range:npt=0-7.712000
      a=length:npt=7.712000
      a=rtpmap:97 mpeg4-generic/32000/2
      a=mimetype:string;"audio/mpeg4-generic"
      a=AvgBitRate:integer;65790
      a=StreamName:string;"hinted audio track"
SETUP
A SETUP request specifies how a single media stream must be transported. This must be done before a PLAY request is sent. The request contains the media stream URL and a transport specifier. This specifier typically includes a local port for receiving RTP data (audio or video), and another for RTCP data (meta information). The server reply usually confirms the chosen parameters, and fills in the missing parts, such as the server's chosen ports. Each media stream must be configured using SETUP before an aggregate play request may be sent.
C->S: SETUP rtsp://example.com/media.mp4/streamid=0 RTSP/1.0
      CSeq: 2
      Transport: RTP/AVP;unicast;client_port=8000-8001

S->C: RTSP/1.0 200 OK
      CSeq: 2
      Transport: RTP/AVP;unicast;client_port=8000-8001;server_port=9000-9001
      Session: 12345678
PLAY
A PLAY request will cause one or all media streams to be played. Play requests can be stacked by sending multiple PLAY requests. The URL may be the aggregate URL (to play all media streams), or a single media stream URL (to play only that stream). A range can be specified. If no range is specified, the stream is played from the beginning and plays to the end, or, if the stream is paused, it is resumed at the point it was paused.
C->S: PLAY rtsp://example.com/media.mp4 RTSP/1.0
      CSeq: 4
      Range: npt=5-20
      Session: 12345678

S->C: RTSP/1.0 200 OK
      CSeq: 4
      Session: 12345678
      RTP-Info: url=rtsp://example.com/media.mp4/streamid=0;seq=9810092;rtptime=3450012
PAUSE
A PAUSE request temporarily halts one or all media streams, so it can later be resumed with a PLAY request. The request contains an aggregate or media stream URL. A range parameter on a PAUSE request specifies when to pause. When the range parameter is omitted, the pause occurs immediately and indefinitely.
C->S: PAUSE rtsp://example.com/media.mp4 RTSP/1.0
      CSeq: 5
      Session: 12345678

S->C: RTSP/1.0 200 OK
      CSeq: 5
      Session: 12345678
RECORD
The RECORD request can be used to send a stream to the server for storage.
TEARDOWN
A TEARDOWN request is used to terminate the session. It stops all media streams and frees all session related data on the server.
C->S: TEARDOWN rtsp://example.com/media.mp4 RTSP/1.0
      CSeq: 6
      Session: 12345678

S->C: RTSP/1.0 200 OK
      CSeq: 6
<think>好的,我现在要处理用户关于RTSP协议的问题。用户提到他们想了解RTSP的实现、使用或文档。首先,我需要回忆RTSP的基本概念和功能,确保信息准确。RTSP主要用于控制实时流媒体,比如播放、暂停等,但传输本身通常通过RTP和RTCP,这点需要明确说明。 接下来,用户可能需要具体的实现步骤或代码示例。我应该考虑常见的实现方式,比如使用开源库如Live555,或者Python的库如python-rtsp-client。需要检查这些库的文档链接是否有效,并确保示例代码的正确性。例如,Python代码示例需要展示如何连接RTSP服务器并获取流,同时提醒用户安装必要的包。 关于文档部分,RFC 2326是RTSP的核心规范,必须引用。此外,FFmpeg和GStreamer的支持文档也很重要,特别是用户可能用这些工具进行流处理。需要确认这些工具的官方文档是否有相关章节。 用户可能还想知道RTSP的应用场景和常见问题。应用场景可以包括IP摄像头、视频会议等。常见问题如防火墙端口、身份验证方式,这些都需要涵盖,并提供解决方法,比如开放端口554或使用摘要认证。 还需要生成相关问题,帮助用户深入探索。例如,RTSP和HTTP的区别,安全机制,与RTP的关系,以及调试工具。这些问题应该覆盖用户可能感兴趣的扩展领域。 最后,确保所有引用正确,特别是站内引用[^1]是否与用户提供的上下文相关。虽然用户提供的引用是关于Darwin Streaming Server的,但需要确认是否直接相关,并在必要时调整引用位置。整个回答结构要清晰,分步骤说明,避免技术错误,同时保持语言自然。</think>### Real-Time Streaming Protocol (RTSP) 实现与使用指南 RTSP 是一种网络控制协议,专为实时流媒体传输(如视频、音频)设计,支持播放、暂停、快进等控制功能。以下是其核心实现与使用要点: --- #### **1. 协议基础** - **功能定位**:RTSP 不直接传输数据,而是通过 RTP(Real-Time Transport Protocol)和 RTCP(RTP Control Protocol)传输媒体流[^1]。其主要职责是控制会话流程。 - **通信方式**:基于 TCP 或 UDP,默认端口为 `554`。 - **典型命令**:`DESCRIBE`(获取媒体信息)、`SETUP`(建立传输通道)、`PLAY`/`PAUSE`(控制播放)。 --- #### **2. 实现方式** ##### **2.1 使用开源库** - **Live555**:C++ 实现的跨平台 RTSP 库,支持服务端与客户端开发 [官方文档](http://www.live555.com/liveMedia/) - **GStreamer RTSP Server**:基于 GStreamer 框架的 RTSP 服务器 [示例代码](https://github.com/GStreamer/gst-rtsp-server) - **Python 实现**(如 `python-rtsp-client`): ```python from rtsp_client import RTSPClient, RTSPAuth client = RTSPClient( rtsp_url="rtsp://example.com/stream", auth=RTSPAuth(username="user", password="pass") ) client.connect() # 获取媒体流描述 response = client.describe() ``` ##### **2.2 简单服务端实现(伪代码)** ```python # 使用 Python socketserver 模拟 RTSP 响应 import socketserver class RTSPHandler(socketserver.BaseRequestHandler): def handle(self): data = self.request.recv(1024).decode() if "DESCRIBE" in data: response = "RTSP/1.0 200 OK\r\nContent-Type: application/sdp\r\n\r\nv=0\r\no=- 0 0 IN IP4 127.0.0.1" self.request.send(response.encode()) server = socketserver.TCPServer(("0.0.0.0", 554), RTSPHandler) server.serve_forever() ``` --- #### **3. 关键文档** - **RFC 2326**:RTSP 协议规范 [RFC 2326 原文](https://tools.ietf.org/html/rfc2326) - **FFmpeg 支持**:通过 `ffmpeg -i rtsp://server/stream` 测试流 [FFmpeg RTSP 指南](https://ffmpeg.org/ffmpeg-protocols.html#rtsp) - **Wireshark 抓包分析**:过滤 `rtsp` 或 `rtp` 协议包以调试交互过程。 --- #### **4. 典型应用场景** - **IP 摄像头监控**:如海康威视、大华摄像头均支持 RTSP 流接入。 - **视频会议系统**:控制媒体会话的启停与参数调整。 - **流媒体服务器**:Darwin Streaming Server(已停维护)曾广泛用于 RTSP 实现。 --- #### **5. 常见问题** - **防火墙配置**:需开放 TCP/UDP 端口 `554` 及 RTP 动态端口范围(如 `30000-40000`)。 - **身份验证**:支持 Basic 或 Digest 认证,需在请求头中携带 `Authorization` 字段。 - **协议兼容性**:部分旧设备仅支持 RTSP over TCP,需明确传输模式。 ---
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