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。