播放 RTSP 视频流
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参考技术A 通过接口调用,获取网络摄像头的 RTSP 推流 URL,需要播放此 RTSP 协议传输的视频流An html5 Flash Video (FLV) Player written in pure javascript without Flash.
FLV 支持格式
动态添加 rtsp | rtmp | hls 拉流代理。只支持 H264 | H265 | AAC | G711 负载
以 rtsp://somedomain.com:554/live/0?token=abcdefg&field=value
通过 流媒体源对应的播放 URL 可知一个 http-flv 的地址 => http://<vhost>/<app>/<streamid>.flv
addStreamProxy response key => <vhost>/<app>/<streamid> => 根据 addStreamProxy response key 得到 http-flv 播放地址 => http://$key.flv
通过 docker 启动 ZLMediaKit 服务
在 ZLMediaKit 配置文件 中详细描述了相关服务器监听端口,flv.js 使用 http 服务器,原视频流是 RTSP 服务器,故端口映射为
通过 Postman 配置文件 导入即可以测试服务
浏览器播放rtsp视频流:3rtsp转webrtc播放
浏览器播放rtsp视频流:3、rtsp转webrtc播放
文章目录
1. 前言
前面我们测试了rtsp转hls方式,发现延迟比较大,不太适合我们的使用需求。接下来我们试一下webrtc的方式看下使用情况。
综合考虑下来,我们最好能找到一个go作为后端,前端兼容性较好的前后端方案来处理webrtc,这样我们就可以结合我们之前的cgo+onvif+gSoap实现方案来获取rtsp流,并且可以根据已经实现的ptz、预置点等功能接口做更多的扩展。
2. rtsp转webRTC
如下是找到的一个比较合适的开源方案,前端使用了jQuery、bootstrap等,后端使用go+gin来实现并将rtsp流解析转换为webRTC协议提供http相关接口给到前端,通过config.json配置rtsp地址和stun地址:
https://github.com/deepch/RTSPtoWebRTC
此外,还带有stun,可以自行配置stun地址,便于进行内网穿透。
初步测试几乎看不出来延迟,符合预期,使用的jQuery+bootstrap+go+gin做的web,也符合我们的项目使用情况。
3. 初步测试结果
结果如下:
4. 结合我们之前的onvif+gSoap+cgo的方案做修改
我们在此项目的基础上,结合我们之前研究的onvif+cgo+gSoap的方案,将onvif获取到的相关数据提供接口到web端,增加ptz、调焦、缩放等功能。
我们在http.go中添加新的post接口:HTTPAPIServerStreamPtz来进行ptz和HTTPAPIServerStreamPreset进行预置点相关操作。
以下是部分代码,没有做太多的优化,也仅仅实现了ptz、调焦和缩放,算是打通了通路,具体项目需要可以再做优化。
4.1 go后端修改
增加了新的接口,并将之前cgo+onvif+gSoap的内容结合了进来,项目整体没有做更多的优化,只是为了演示,提供一个思路:
http.go(增加了两个post接口ptz和preset,采用cgo方式处理):
package main
/*
#cgo CFLAGS: -I ./ -I /usr/local/
#cgo LDFLAGS: -L ./build -lc_onvif_static -lpthread -ldl -lssl -lcrypto
#include "client.h"
#include "malloc.h"
*/
import "C"
import (
"encoding/json"
"fmt"
"log"
"net/http"
"os"
"sort"
"strconv"
"time"
"unsafe"
"github.com/deepch/vdk/av"
webrtc "github.com/deepch/vdk/format/webrtcv3"
"github.com/gin-gonic/gin"
)
type JCodec struct
Type string
func serveHTTP()
gin.SetMode(gin.ReleaseMode)
router := gin.Default()
router.Use(CORSMiddleware())
if _, err := os.Stat("./web"); !os.IsNotExist(err)
router.LoadHTMLGlob("web/templates/*")
router.GET("/", HTTPAPIServerIndex)
router.GET("/stream/player/:uuid", HTTPAPIServerStreamPlayer)
router.POST("/stream/receiver/:uuid", HTTPAPIServerStreamWebRTC)
//增加新的post接口
router.POST("/stream/ptz/", HTTPAPIServerStreamPtz)
router.POST("/stream/preset/", HTTPAPIServerStreamPreset)
router.GET("/stream/codec/:uuid", HTTPAPIServerStreamCodec)
router.POST("/stream", HTTPAPIServerStreamWebRTC2)
router.StaticFS("/static", http.Dir("web/static"))
err := router.Run(Config.Server.HTTPPort)
if err != nil
log.Fatalln("Start HTTP Server error", err)
//HTTPAPIServerIndex index
func HTTPAPIServerIndex(c *gin.Context)
_, all := Config.list()
if len(all) > 0
c.Header("Cache-Control", "no-cache, max-age=0, must-revalidate, no-store")
c.Header("Access-Control-Allow-Origin", "*")
c.Redirect(http.StatusMovedPermanently, "stream/player/"+all[0])
else
c.HTML(http.StatusOK, "index.tmpl", gin.H
"port": Config.Server.HTTPPort,
"version": time.Now().String(),
)
//HTTPAPIServerStreamPlayer stream player
func HTTPAPIServerStreamPlayer(c *gin.Context)
_, all := Config.list()
sort.Strings(all)
c.HTML(http.StatusOK, "player.tmpl", gin.H
"port": Config.Server.HTTPPort,
"suuid": c.Param("uuid"),
"suuidMap": all,
"version": time.Now().String(),
)
//HTTPAPIServerStreamCodec stream codec
func HTTPAPIServerStreamCodec(c *gin.Context)
if Config.ext(c.Param("uuid"))
Config.RunIFNotRun(c.Param("uuid"))
codecs := Config.coGe(c.Param("uuid"))
if codecs == nil
return
var tmpCodec []JCodec
for _, codec := range codecs
if codec.Type() != av.H264 && codec.Type() != av.PCM_ALAW && codec.Type() != av.PCM_MULAW && codec.Type() != av.OPUS
log.Println("Codec Not Supported WebRTC ignore this track", codec.Type())
continue
if codec.Type().IsVideo()
tmpCodec = append(tmpCodec, JCodecType: "video")
else
tmpCodec = append(tmpCodec, JCodecType: "audio")
b, err := json.Marshal(tmpCodec)
if err == nil
_, err = c.Writer.Write(b)
if err != nil
log.Println("Write Codec Info error", err)
return
//HTTPAPIServerStreamWebRTC stream video over WebRTC
func HTTPAPIServerStreamWebRTC(c *gin.Context)
if !Config.ext(c.PostForm("suuid"))
log.Println("Stream Not Found")
return
Config.RunIFNotRun(c.PostForm("suuid"))
codecs := Config.coGe(c.PostForm("suuid"))
if codecs == nil
log.Println("Stream Codec Not Found")
return
var AudioOnly bool
if len(codecs) == 1 && codecs[0].Type().IsAudio()
AudioOnly = true
muxerWebRTC := webrtc.NewMuxer(webrtc.OptionsICEServers: Config.GetICEServers(), ICEUsername: Config.GetICEUsername(), ICECredential: Config.GetICECredential(), PortMin: Config.GetWebRTCPortMin(), PortMax: Config.GetWebRTCPortMax())
answer, err := muxerWebRTC.WriteHeader(codecs, c.PostForm("data"))
if err != nil
log.Println("WriteHeader", err)
return
_, err = c.Writer.Write([]byte(answer))
if err != nil
log.Println("Write", err)
return
go func()
cid, ch := Config.clAd(c.PostForm("suuid"))
defer Config.clDe(c.PostForm("suuid"), cid)
defer muxerWebRTC.Close()
var videoStart bool
noVideo := time.NewTimer(10 * time.Second)
for
select
case <-noVideo.C:
log.Println("noVideo")
return
case pck := <-ch:
if pck.IsKeyFrame || AudioOnly
noVideo.Reset(10 * time.Second)
videoStart = true
if !videoStart && !AudioOnly
continue
err = muxerWebRTC.WritePacket(pck)
if err != nil
log.Println("WritePacket", err)
return
()
func HTTPAPIServerStreamPtz(c *gin.Context)
action := c.PostForm("action")
direction, err := strconv.Atoi(action)
if err != nil
log.Println(err)
return
var soap C.P_Soap
soap = C.new_soap(soap)
username := C.CString("admin")
password := C.CString("admin")
serviceAddr := C.CString("http://40.40.40.101:80/onvif/device_service")
C.get_device_info(soap, username, password, serviceAddr)
mediaAddr := [200]C.char
C.get_capabilities(soap, username, password, serviceAddr, &mediaAddr[0])
profileToken := [200]C.char
C.get_profiles(soap, username, password, &profileToken[0], &mediaAddr[0])
videoSourceToken := [200]C.char
C.get_video_source(soap, username, password, &videoSourceToken[0], &mediaAddr[0])
switch direction
case 0:
break
case 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11:
C.ptz(soap, username, password, C.int(direction), C.float(0.5), &profileToken[0], &mediaAddr[0])
case 12, 13, 14:
C.focus(soap, username, password, C.int(direction), C.float(0.5), &videoSourceToken[0], &mediaAddr[0])
default:
fmt.Println("Unknown direction.")
C.del_soap(soap)
C.free(unsafe.Pointer(username))
C.free(unsafe.Pointer(password))
C.free(unsafe.Pointer(serviceAddr))
c.JSON(http.StatusOK, gin.H"message":"success")
func HTTPAPIServerStreamPreset(c *gin.Context)
var soap C.P_Soap
soap = C.new_soap(soap)
username := C.CString("admin")
password := C.CString("admin")
serviceAddr := C.CString("http://40.40.40.101:80/onvif/device_service")
C.get_device_info(soap, username, password, serviceAddr)
mediaAddr := [200]C.char
C.get_capabilities(soap, username, password, serviceAddr, &mediaAddr[0])
profileToken := [200]C.char
C.get_profiles(soap, username, password, &profileToken[0], &mediaAddr[0])
videoSourceToken := [200]C.char
C.get_video_source(soap, username, password, &videoSourceToken[0], &mediaAddr[0])
action := c.PostForm("action")
presetAction, err := strconv.Atoi(action)
if err != nil
log.Println(err)
return
fmt.Println("请输入数字进行preset,1-4分别代表查询、设置、跳转、删除预置点;退出输入0:")
switch presetAction
case 0:
break
case 1:
C.preset(soap, username, password, C.int(presetAction), nil, nil, &profileToken[0], &mediaAddr[0])
case 2:
fmt.Println("请输入要设置的预置点token信息:")
presentToken := ""
_, _ = fmt.Scanln(&presentToken)
fmt.Println("请输入要设置的预置点name信息长度不超过200:")
presentName := ""
_, _ = fmt.Scanln(&presentName)
C.preset(soap, username, password, C.int(presetAction), C.CString(presentToken), C.CString(presentName), &profileToken[0], &mediaAddr[0])
case 3:
fmt.Println("请输入要跳转的预置点token信息:")
presentToken := ""
_, _ = fmt.Scanln(&presentToken)
C.preset(soap, username, password, C.int(presetAction), C.CString(presentToken), nil, &profileToken[0], &mediaAddr[0])
case 4:
fmt.Println("请输入要删除的预置点token信息:")
presentToken := ""
_, _ = fmt.Scanln(&presentToken)
C.preset(soap, username, password, C.int(presetAction), C.CString(presentToken), nil, &profileToken[0], &mediaAddr[0])
default:
fmt.Println("unknown present action.")
break
C.del_soap(soap)
C.free(unsafe.Pointer(username))
C.free(unsafe.Pointer(password))
C.free(unsafe.Pointer(serviceAddr))
c.JSON(http.StatusOK, gin.H"message":"success")
func CORSMiddleware() gin.HandlerFunc
return func(c *gin.Context)
c.Header("Access-Control-Allow-Origin", "*")
c.Header("Access-Control-Allow-Credentials", "true")
c.Header("Access-Control-Allow-Headers", "Origin, X-Requested-With, Content-Type, Accept, Authorization, x-access-token")
c.Header("Access-Control-Expose-Headers", "Content-Length, Access-Control-Allow-Origin, Access-Control-A以上是关于播放 RTSP 视频流的主要内容,如果未能解决你的问题,请参考以下文章