从锁定屏幕接听电话时,Callkit 和 Webrtc 没有音频
Callkit and Webrtc no audio when accept call from locked screen
我试图让 callkit 在来电时与 webrtc 一起工作,但是当我接听电话并从锁定屏幕接听时,直到我 运行 应用程序处于前台模式时才会有声音。我已经配置了音频会话向 RTCAudioSession 发送通知,但它不起作用。你有一些解决方法吗?
func configureAudioSession() {
let sharedSession = AVAudioSession.sharedInstance()
do {
try sharedSession.setCategory(AVAudioSessionCategoryPlayAndRecord, mode: AVAudioSessionModeVideoChat, options: .mixWithOthers)
try sharedSession.setMode(AVAudioSessionModeVideoChat)
// try sharedSession.setAggregatedIOPreference(AVAudioSessionIOType.aggregated)
} catch {
debugPrint("Failed to configure `AVAudioSession`")
}
}
func handleIncomingCall(spaceName:String) {
if callUUID != nil {
oldCallUUID = callUUID
}
callUUID = UUID()
print("CallManager handle uuid = \(callUUID?.description)")
let update = CXCallUpdate()
update.hasVideo = true
update.remoteHandle = CXHandle(type: .generic, value: spaceName)
self.configureAudioSession()
provider?.reportNewIncomingCall(with: callUUID!, update: update, completion: { error in
print("CallManager report new incoming call completion")
})
}
func provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession) {
print("CallManager didActivate")
RTCAudioSession.sharedInstance().audioSessionDidActivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = true
self.callDelegate?.callIsAnswered()
}
func provider(_ provider: CXProvider, didDeactivate audioSession: AVAudioSession) {
print("CallManager didDeactivate")
RTCAudioSession.sharedInstance().audioSessionDidDeactivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = false
}
您的测试 iOS 版本是什么 iPhone?
好的,我找到问题的原因了。在 IOS 12 中,webrtc 存在问题,当您从锁定屏幕启动 webrtc 并尝试访问相机时 - 输出音量中断,因此解决方案是检查屏幕是否处于活动状态,以及是否not - 不请求本地 RTCVideoTrack 并将其添加到您的 RTCStream 中。
请注意,我分享了我的代码及其关于我的需要,我分享以供参考。你需要根据你的需要改变它。
当您收到 voip 通知时,创建新的 webrtc 处理事件 class,以及
将这两行添加到代码块,因为从 voip 通知启用音频会话失败
RTCAudioSession.sharedInstance().useManualAudio = true
RTCAudioSession.sharedInstance().isAudioEnabled = false
didReceive 方法;
func pushRegistry(_ registry: PKPushRegistry, didReceiveIncomingPushWith payload: PKPushPayload, for type: PKPushType, completion: @escaping () -> Void) {
let state = UIApplication.shared.applicationState
if(payload.dictionaryPayload["hangup"] == nil && state != .active
){
Globals.voipPayload = payload.dictionaryPayload as! [String:Any] // I pass parameters to Webrtc handler via Global singleton to create answer according to sdp sent by payload.
RTCAudioSession.sharedInstance().useManualAudio = true
RTCAudioSession.sharedInstance().isAudioEnabled = false
Globals.sipGateway = SipGateway() // my Webrtc and Janus gateway handler class
Globals.sipGateway?.configureCredentials(true) // I check janus gateway credentials stored in Shared preferences and initiate websocket connection and create peerconnection
to my janus gateway which is signaling server for my environment
initProvider() //Crating callkit provider
self.update.remoteHandle = CXHandle(type: .generic, value:String(describing: payload.dictionaryPayload["caller_id"]!))
Globals.callId = UUID()
let state = UIApplication.shared.applicationState
Globals.provider.reportNewIncomingCall(with:Globals.callId , update: self.update, completion: { error in
})
}
}
func initProvider(){
let config = CXProviderConfiguration(localizedName: "ulakBEL")
config.iconTemplateImageData = UIImage(named: "ulakbel")!.pngData()
config.ringtoneSound = "ringtone.caf"
// config.includesCallsInRecents = false;
config.supportsVideo = false
Globals.provider = CXProvider(configuration:config )
Globals.provider.setDelegate(self, queue: nil)
update = CXCallUpdate()
update.hasVideo = false
update.supportsDTMF = true
}
修改您的 didActivate 和 didDeActive 委托函数,如下所示,
func provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession) {
print("CallManager didActivate")
RTCAudioSession.sharedInstance().audioSessionDidActivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = true
// self.callDelegate?.callIsAnswered()
}
func provider(_ provider: CXProvider, didDeactivate audioSession: AVAudioSession) {
print("CallManager didDeactivate")
RTCAudioSession.sharedInstance().audioSessionDidDeactivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = false
}
在 Webrtc 处理程序中 class 配置媒体发送器和音频会话
private func createPeerConnection(webRTCCallbacks:PluginHandleWebRTCCallbacksDelegate) {
let rtcConfig = RTCConfiguration.init()
rtcConfig.iceServers = server.iceServers
rtcConfig.bundlePolicy = RTCBundlePolicy.maxBundle
rtcConfig.rtcpMuxPolicy = RTCRtcpMuxPolicy.require
rtcConfig.continualGatheringPolicy = .gatherContinually
rtcConfig.sdpSemantics = .planB
let constraints = RTCMediaConstraints(mandatoryConstraints: nil,
optionalConstraints: ["DtlsSrtpKeyAgreement":kRTCMediaConstraintsValueTrue])
pc = sessionFactory.peerConnection(with: rtcConfig, constraints: constraints, delegate: nil)
self.createMediaSenders()
self.configureAudioSession()
if webRTCCallbacks.getJsep() != nil{
handleRemoteJsep(webrtcCallbacks: webRTCCallbacks)
}
}
mediaSenders;
private func createMediaSenders() {
let streamId = "stream"
// Audio
let audioTrack = self.createAudioTrack()
self.pc.add(audioTrack, streamIds: [streamId])
// Video
/* let videoTrack = self.createVideoTrack()
self.localVideoTrack = videoTrack
self.peerConnection.add(videoTrack, streamIds: [streamId])
self.remoteVideoTrack = self.peerConnection.transceivers.first { [=14=].mediaType == .video }?.receiver.track as? RTCVideoTrack
// Data
if let dataChannel = createDataChannel() {
dataChannel.delegate = self
self.localDataChannel = dataChannel
}*/
}
private func createAudioTrack() -> RTCAudioTrack {
let audioConstrains = RTCMediaConstraints(mandatoryConstraints: nil, optionalConstraints: nil)
let audioSource = sessionFactory.audioSource(with: audioConstrains)
let audioTrack = sessionFactory.audioTrack(with: audioSource, trackId: "audio0")
return audioTrack
}
音频会话;
private func configureAudioSession() {
self.rtcAudioSession.lockForConfiguration()
do {
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
try self.rtcAudioSession.setMode(AVAudioSession.Mode.voiceChat.rawValue)
} catch let error {
debugPrint("Error changeing AVAudioSession category: \(error)")
}
self.rtcAudioSession.unlockForConfiguration()
}
请考虑一下,因为我使用回调和委托代码包括委托和回调块。你可以相应地忽略它们!!
供参考您还可以查看此处的示例 link
我试图让 callkit 在来电时与 webrtc 一起工作,但是当我接听电话并从锁定屏幕接听时,直到我 运行 应用程序处于前台模式时才会有声音。我已经配置了音频会话向 RTCAudioSession 发送通知,但它不起作用。你有一些解决方法吗?
func configureAudioSession() {
let sharedSession = AVAudioSession.sharedInstance()
do {
try sharedSession.setCategory(AVAudioSessionCategoryPlayAndRecord, mode: AVAudioSessionModeVideoChat, options: .mixWithOthers)
try sharedSession.setMode(AVAudioSessionModeVideoChat)
// try sharedSession.setAggregatedIOPreference(AVAudioSessionIOType.aggregated)
} catch {
debugPrint("Failed to configure `AVAudioSession`")
}
}
func handleIncomingCall(spaceName:String) {
if callUUID != nil {
oldCallUUID = callUUID
}
callUUID = UUID()
print("CallManager handle uuid = \(callUUID?.description)")
let update = CXCallUpdate()
update.hasVideo = true
update.remoteHandle = CXHandle(type: .generic, value: spaceName)
self.configureAudioSession()
provider?.reportNewIncomingCall(with: callUUID!, update: update, completion: { error in
print("CallManager report new incoming call completion")
})
}
func provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession) {
print("CallManager didActivate")
RTCAudioSession.sharedInstance().audioSessionDidActivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = true
self.callDelegate?.callIsAnswered()
}
func provider(_ provider: CXProvider, didDeactivate audioSession: AVAudioSession) {
print("CallManager didDeactivate")
RTCAudioSession.sharedInstance().audioSessionDidDeactivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = false
}
您的测试 iOS 版本是什么 iPhone?
好的,我找到问题的原因了。在 IOS 12 中,webrtc 存在问题,当您从锁定屏幕启动 webrtc 并尝试访问相机时 - 输出音量中断,因此解决方案是检查屏幕是否处于活动状态,以及是否not - 不请求本地 RTCVideoTrack 并将其添加到您的 RTCStream 中。
请注意,我分享了我的代码及其关于我的需要,我分享以供参考。你需要根据你的需要改变它。
当您收到 voip 通知时,创建新的 webrtc 处理事件 class,以及 将这两行添加到代码块,因为从 voip 通知启用音频会话失败
RTCAudioSession.sharedInstance().useManualAudio = true
RTCAudioSession.sharedInstance().isAudioEnabled = false
didReceive 方法;
func pushRegistry(_ registry: PKPushRegistry, didReceiveIncomingPushWith payload: PKPushPayload, for type: PKPushType, completion: @escaping () -> Void) {
let state = UIApplication.shared.applicationState
if(payload.dictionaryPayload["hangup"] == nil && state != .active
){
Globals.voipPayload = payload.dictionaryPayload as! [String:Any] // I pass parameters to Webrtc handler via Global singleton to create answer according to sdp sent by payload.
RTCAudioSession.sharedInstance().useManualAudio = true
RTCAudioSession.sharedInstance().isAudioEnabled = false
Globals.sipGateway = SipGateway() // my Webrtc and Janus gateway handler class
Globals.sipGateway?.configureCredentials(true) // I check janus gateway credentials stored in Shared preferences and initiate websocket connection and create peerconnection
to my janus gateway which is signaling server for my environment
initProvider() //Crating callkit provider
self.update.remoteHandle = CXHandle(type: .generic, value:String(describing: payload.dictionaryPayload["caller_id"]!))
Globals.callId = UUID()
let state = UIApplication.shared.applicationState
Globals.provider.reportNewIncomingCall(with:Globals.callId , update: self.update, completion: { error in
})
}
}
func initProvider(){
let config = CXProviderConfiguration(localizedName: "ulakBEL")
config.iconTemplateImageData = UIImage(named: "ulakbel")!.pngData()
config.ringtoneSound = "ringtone.caf"
// config.includesCallsInRecents = false;
config.supportsVideo = false
Globals.provider = CXProvider(configuration:config )
Globals.provider.setDelegate(self, queue: nil)
update = CXCallUpdate()
update.hasVideo = false
update.supportsDTMF = true
}
修改您的 didActivate 和 didDeActive 委托函数,如下所示,
func provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession) {
print("CallManager didActivate")
RTCAudioSession.sharedInstance().audioSessionDidActivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = true
// self.callDelegate?.callIsAnswered()
}
func provider(_ provider: CXProvider, didDeactivate audioSession: AVAudioSession) {
print("CallManager didDeactivate")
RTCAudioSession.sharedInstance().audioSessionDidDeactivate(audioSession)
RTCAudioSession.sharedInstance().isAudioEnabled = false
}
在 Webrtc 处理程序中 class 配置媒体发送器和音频会话
private func createPeerConnection(webRTCCallbacks:PluginHandleWebRTCCallbacksDelegate) {
let rtcConfig = RTCConfiguration.init()
rtcConfig.iceServers = server.iceServers
rtcConfig.bundlePolicy = RTCBundlePolicy.maxBundle
rtcConfig.rtcpMuxPolicy = RTCRtcpMuxPolicy.require
rtcConfig.continualGatheringPolicy = .gatherContinually
rtcConfig.sdpSemantics = .planB
let constraints = RTCMediaConstraints(mandatoryConstraints: nil,
optionalConstraints: ["DtlsSrtpKeyAgreement":kRTCMediaConstraintsValueTrue])
pc = sessionFactory.peerConnection(with: rtcConfig, constraints: constraints, delegate: nil)
self.createMediaSenders()
self.configureAudioSession()
if webRTCCallbacks.getJsep() != nil{
handleRemoteJsep(webrtcCallbacks: webRTCCallbacks)
}
}
mediaSenders;
private func createMediaSenders() {
let streamId = "stream"
// Audio
let audioTrack = self.createAudioTrack()
self.pc.add(audioTrack, streamIds: [streamId])
// Video
/* let videoTrack = self.createVideoTrack()
self.localVideoTrack = videoTrack
self.peerConnection.add(videoTrack, streamIds: [streamId])
self.remoteVideoTrack = self.peerConnection.transceivers.first { [=14=].mediaType == .video }?.receiver.track as? RTCVideoTrack
// Data
if let dataChannel = createDataChannel() {
dataChannel.delegate = self
self.localDataChannel = dataChannel
}*/
}
private func createAudioTrack() -> RTCAudioTrack {
let audioConstrains = RTCMediaConstraints(mandatoryConstraints: nil, optionalConstraints: nil)
let audioSource = sessionFactory.audioSource(with: audioConstrains)
let audioTrack = sessionFactory.audioTrack(with: audioSource, trackId: "audio0")
return audioTrack
}
音频会话;
private func configureAudioSession() {
self.rtcAudioSession.lockForConfiguration()
do {
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
try self.rtcAudioSession.setMode(AVAudioSession.Mode.voiceChat.rawValue)
} catch let error {
debugPrint("Error changeing AVAudioSession category: \(error)")
}
self.rtcAudioSession.unlockForConfiguration()
}
请考虑一下,因为我使用回调和委托代码包括委托和回调块。你可以相应地忽略它们!!
供参考您还可以查看此处的示例 link