基于SRS服务器实现Android-Web端视频通话（2）：Android端从SRS服务器拉取WebRTC流

基于SRS服务器实现Android-Web端视频通话（1）：SRS服务器启用HTTPS
基于SRS服务器实现Android-Web端视频通话（2）：Android端从SRS服务器拉取WebRTC流
基于SRS服务器实现Android-Web端视频通话（3）：Android端向SRS服务器推送WebRTC流

实现效果

引库

implementation 'org.webrtc:google-webrtc:1.0.32006'

其他版本，详见

拉流流程

createPeerConnectionFactory -> createPeerConnection -> createOffer -> setLocalDescription(OFFER) -> get remote sdp(network requset) -> setRemoteDescription(ANSWER)

代码实现

初始化

//加载并初始化 WebRTC，在创建 PeerConnectionFactory 之前必须至少调用一次
PeerConnectionFactory.initialize(
    PeerConnectionFactory.InitializationOptions
        .builder(applicationContext).createInitializationOptions()
)

private val eglBaseContext = EglBase.create().eglBaseContext

createPeerConnectionFactory

private lateinit var peerConnectionFactory: PeerConnectionFactory
...
//一些默认初始化配置即可
val options = PeerConnectionFactory.Options()
val encoderFactory = DefaultVideoEncoderFactory(eglBaseContext, true, true)
val decoderFactory = DefaultVideoDecoderFactory(eglBaseContext)
peerConnectionFactory = PeerConnectionFactory.builder()
    setOptions(options)
    .setVideoEncoderFactory(encoderFactory)
    .setVideoDecoderFactory(decoderFactory)
    .createPeerConnectionFactory()
...

createPeerConnection

val rtcConfig = PeerConnection.RTCConfiguration(emptyList())
/*
 <p>For users who wish to send multiple audio/video streams and need to stay interoperable with legacy WebRTC implementations, specify PLAN_B.
 <p>For users who wish to send multiple audio/video streams and/or wish to use the new RtpTransceiver API, specify UNIFIED_PLAN.
 */
//使用PeerConnection.SdpSemantics.UNIFIED_PLAN
rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN
val peerConnection = peerConnectionFactory.createPeerConnection(
    rtcConfig,
    object : PeerConnectionObserver() {
    	/**
    	 * Triggered when media is received on a new stream from remote peer.
    	 * 当收到远端媒体流时调用
    	 */
        override fun onAddStream(mediaStream: MediaStream?) {
            super.onAddStream(mediaStream)
            mediaStream?.let {
            	//如果有视频轨。
                if (it.videoTracks.isEmpty().not()) {
                    it.videoTracks[0].addSink(mBinding.svr)
                }
            }
        }
    })?.apply {
    //addTransceiver()调用顺序会影响到生成sdp信息中video、audio信息顺序，对应关系，下面会提到；
    //接收视频，指定仅接收即可
    addTransceiver(
        MediaStreamTrack.MediaType.MEDIA_TYPE_VIDEO,
        RtpTransceiver.RtpTransceiverInit(RtpTransceiver.RtpTransceiverDirection.RECV_ONLY)
    )
    //接收音频，指定仅接收即可
    addTransceiver(
        MediaStreamTrack.MediaType.MEDIA_TYPE_AUDIO,
        RtpTransceiver.RtpTransceiverInit(RtpTransceiver.RtpTransceiverDirection.RECV_ONLY)
    )
}

createOffer && setLocalDescription

peerConnection.createOffer(object : SdpAdapter("createOffer") {
    override fun onCreateSuccess(description: SessionDescription?) {
        super.onCreateSuccess(description)
        description?.let {
            if (it.type == SessionDescription.Type.OFFER) {     
            	peerConnection.setLocalDescription(SdpAdapter("setLocalDescription"), it)
            	//这个offerSdp将用于向SRS服务进行网络请求
				val offerSdp = it.description
				getRemoteSdp(offerSdp)             
            }
        }
    }
}, MediaConstraints())

get remote sdp(netword requset)

基本配置，根据自己实际情况进行调整

object Constant {
    /**
     * SRS服务器IP
     */
    const val SRS_SERVER_IP = "192.168.2.91"

    /**
     * SRS服务http请求端口，默认1985
     */
    const val SRS_SERVER_HTTP_PORT = "1985"

    /**
     * SRS服务https请求端口，默认1990
     */
    const val SRS_SERVER_HTTPS_PORT = "1990"

    const val SRS_SERVER_HTTP = "$SRS_SERVER_IP:$SRS_SERVER_HTTP_PORT"

    const val SRS_SERVER_HTTPS = "$SRS_SERVER_IP:$SRS_SERVER_HTTPS_PORT"
}

Request Body (application/json)

data class SrsRequestBean(
    /**
     * [PeerConnection.createOffer]返回的sdp
     */
    @Json(name = "sdp")
    val sdp: String?,
    /**
     * 拉取的WebRTC流地址
     */
    @Json(name = "streamurl")
    val streamUrl: String?
)

Response Body (application/json)

data class SrsResponseBean(
    /**
     * 0：成功
     */
    @Json(name = "code")
    val code: Int,
    /**
     * 用于设置[PeerConnection.setRemoteDescription]
     */
    @Json(name = "sdp") val sdp: String?,
    @Json(name = "server")
    val server: String?,
    @Json(name = "sessionid")
    val sessionId: String?
)

网络请求地址
http请求：http://ip:port/rtc/v1/play/
https请求：https://ip:port/rtc/v1/play/
Method：POST

在Android P(28)系统的设备上，禁止应用使用的是非加密的明文流量的HTTP 网络请求。

retrofit事例

interface ApiService {

    @POST("/rtc/v1/play/")
    suspend fun play(@Body body: SrsRequestBean): SrsResponseBean
}

getRemoteSdp

private fun getRemoteSdp(offerSdp: String){
	//webrtc流地址
	val webrtcUrl="webrtc://${Constant.SRS_SERVER_IP}/live/livestream"
    val srsBean = SrsRequestBean(offerSdp, webrtcUrl)
    lifecycleScope.launch {
        val result = try {
            withContext(Dispatchers.IO) {
                retrofitClient.apiService.play(srsBean)
            }
        } catch (e: Exception) {
            println("网络请求出错：${e.printStackTrace()}")
            toastError("网络请求出错：${e.printStackTrace()}")
            null
        }

        result?.let { bean ->
            if (bean.code == 0) {
                println("网络请求成功，code：${bean.code}")
				setRemoteDescription(bean.sdp)
            } else {
                println("网络请求失败，code：${bean.code}")
            }
        }
    }
}

setRemoteDescription

private fun setRemoteDescription(answerSdp: String){
	val remoteSdp = SessionDescription(SessionDescription.Type.ANSWER, /*关键点*/answerSdp)
	//注意这一步，可能会报错：Failed to set remote answer sdp: The order of m-lines in answer doesn't match order in offer. Rejecting answer.
	peerConnection.setRemoteDescription(SdpAdapter("setRemoteDescription"), remoteSdp)
}

如果你遇到这个错误：
Failed to set remote answer sdp: The order of m-lines in answer doesn’t match order in offer. Rejecting answer.

可以看下我的另外一篇博客。
具体原因就是offer中sdp的video、audio和answer中sdp的video、audio顺序不一致，详见#3179。
解决方案目前大致有三种：
1、升级SRS服务器版本，截止2022-09-16，升级到v4.0.265及以上，即可解决，云服务器这个版本做了兼容处理；
2、如果无法升级SRS，可以调整上面提到的调整**addTransceiver()**调用顺序，确保顺序与SRS接口返回的sdp中video、audio顺序一致；

ps: 上面两种方案解决方式还要非常感谢SRS-杨成立指点，十分感谢！

3、使用博客中第一种原因的解决方式，我们需要手动调换下位置（当然这个方法也可以作为最后的保护手段）：

/**
 * 转换AnswerSdp
 * @param offerSdp offerSdp：创建offer时生成的sdp
 * @param answerSdp answerSdp：网络请求srs服务器返回的sdp
 * @return 转换后的AnswerSdp
 */
private fun convertAnswerSdp(offerSdp: String, answerSdp: String?): String {
	if (answerSdp.isNullOrBlank()){
		return ""
    }
    val indexOfOfferVideo = offerSdp.indexOf("m=video")
    val indexOfOfferAudio = offerSdp.indexOf("m=audio")
    if (indexOfOfferVideo == -1 || indexOfOfferAudio == -1) {
        return answerSdp
    }
    val indexOfAnswerVideo = answerSdp.indexOf("m=video")
    val indexOfAnswerAudio = answerSdp.indexOf("m=audio")
    if (indexOfAnswerVideo == -1 || indexOfAnswerAudio == -1) {
        return answerSdp
    }

    val isFirstOfferVideo = indexOfOfferVideo < indexOfOfferAudio
    val isFirstAnswerVideo = indexOfAnswerVideo < indexOfAnswerAudio
    return if (isFirstOfferVideo == isFirstAnswerVideo) {
        //顺序一致
        answerSdp
    } else {
        //需要调换顺序
        buildString {
            append(answerSdp.substring(0, indexOfAnswerVideo.coerceAtMost(indexOfAnswerAudio)))
            append(
                answerSdp.substring(
                    indexOfAnswerVideo.coerceAtLeast(indexOfOfferVideo),
                    answerSdp.length
                )
            )
            append(
                answerSdp.substring(
                    indexOfAnswerVideo.coerceAtMost(indexOfAnswerAudio),
                    indexOfAnswerVideo.coerceAtLeast(indexOfOfferVideo)
                )
            )
        }
    }
}

修改方法：

private fun setRemoteDescription(offerSdp: String, answerSdp: String){
	val remoteSdp = SessionDescription(SessionDescription.Type.ANSWER, /*关键点*/convertAnswerSdp(offerSdp, answerSdp))
	peerConnection.setRemoteDescription(SdpAdapter("setRemoteDescription"), remoteSdp)
}

关闭

释放资源，避免内存泄漏

mBinding.svr.release()
peerConnection?.dispose()
peerConnectionFactory.dispose()

至此，拉流播放流程结束。如有错误欢迎指正。

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