srs webrtc服务p2p案例信令分析

srs版本为v4.0-r3

概述

srs的webrtc功能由三个组件共同完成：srs流媒体服务(默认端口1985)、3rdparty/signaling下的信令服务（默认端口1989）以及3rdparty/httpx-static的http/https代理服务组成。httpx-static监听http(80)及https（443）端口，并将/sig请求转发给信令服务，将/rtc请求转发给srs流媒体服务。

房间信令部分

信令连接类型为websocket，连接请求的url类似下面的样子

ws://xx.com/sig/v1/rtc?room=live&display=234e488

room为房间号，display为用户名。默认情况下，这个请求首先到达httpx-static服务，再由httpx-static服务转请求转发到signaling服务进行处理。

加入房间请求（join）

{"tid":"81515ac","msg":{"action":"join","room":"10098","display":"85e63c6"}}

加入房间返回

{
	"tid": "81515ac",
	"msg": {
		"action": "join",
		"room": "live",
		"self": {
			"display": "6e1e3d8",
			"publishing": false
		},
		"participants": [{
			"display": "6e1e3d8",
			"publishing": false
		}]
	}
}

当房间有其他用户时，participants字段里将有其他用户的信息，其返回如下：

{
	"tid": "81515ac",
	"msg": {
		"action": "join",
		"room": "live",
		"self": {
			"display": "12789",
			"publishing": false
		},
		"participants": [{
			"display": "6e1e3d8",
			"publishing": true
		}, {
			"display": "12789",
			"publishing": false
		}]
	}
}

发布流请求

{"tid":"2f65662","msg":{"action":"publish","room":"live","display":"337ecd3"}}

发布流返回

{"tid":"2f65662","msg":null}

Notify通知

当房间有其他人join的时候，会收到如下notify:

{
	"msg": {
		"action": "notify",
		"event": "join",
		"room": "live",
		"self": {
			"display": "337ecd3",
			"publishing": true
		},
		"peer": {
			"display": "6f22ee1",
			"publishing": false
		},
		"participants": [{
			"display": "337ecd3",
			"publishing": true
		}, {
			"display": "6f22ee1",
			"publishing": false
		}]
	}
}

当房间有其他人publish流的时候，会收到如下notify:

{
	"msg": {
		"action": "notify",
		"event": "publish",
		"room": "live",
		"self": {
			"display": "337ecd3",
			"publishing": true
		},
		"peer": {
			"display": "6f22ee1",
			"publishing": true
		},
		"participants": [{
			"display": "337ecd3",
			"publishing": true
		}, {
			"display": "6f22ee1",
			"publishing": true
		}]
	}
}

推拉流信令部分

推流信令由srs流媒体服务进行处理，默认端口为1985。

推流信令

通过http post发送

method: POST
url: /rtc/v1/publish/
ContentType: application/json

一个可能的content如下：

{
    "api": "http://xxx.com:1985/rtc/v1/publish/",
    "tid": "905b653",
    "streamurl": "webrtc://xxxx.com/live/337ecd3null",
    "clientip": null,
    "sdp": "v=0\r\no=- 3779124144089994960 ...\r\n"
}

上面sdp就是webrtc的sdp，这些进行了省略。

srs.sdk.js中提供的示例代码片段如下：

var data = {
                api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
                clientip: null, sdp: offer.sdp
            };
console.log("Generated offer: ", data);
 
$.ajax({
     type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
     contentType: 'application/json', dataType: 'json'
})

下面是返回示例：

{
    "code": 0,
    "server": "vid-a8mj4jz",
    "sdp": "v=0\r\no=SRS/4.0.265(Leo) 94612519010016 2 IN IP4 0.0.0.0\r\ns=SRSPublishSession\r\n...",
    "sessionid": "t95zu036:l4Ai",
    "simulator": "http://xxxx.com:1985/rtc/v1/nack/"
}

拉流信令

方法是OPTIONS???

method: POST
url: /rtc/v1/play/
ContentType: application/json

content:

{
    "api": "http://xxxx.com:1985/rtc/v1/play/",
    "tid": "4b84522",
    "streamurl": "webrtc://xxxx.com/live/6f22ee1null",
    "clientip": null,
    "sdp": "v=0\r\no=- 2143530685089536534 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1...\r\n"
}

返回示例如下：

{
    "code": 0,
    "server": "vid-a8mj4jz",
    "sdp": "v=0\r\no=SRS/4.0.265(Leo) 94612519489568 2 IN IP4 0.0.0.0\r\ns=SRSPlaySession\r\nt=0 0\r\na=ice-lite...\r\n",
    "sessionid": "637a69d0:6L8Z",
    "simulator": "http://xxxx.com:1985/rtc/v1/nack/"
}