Solution for commentators. Ingest RTMP and display via WEBRTC. Comment via WebRTC and mix it to hls

[VitalyZaikin]

Hi there! I'm new here.
I need to get an RTMP stream from the studio and display it via WebRTC. The commentator will view this stream via WebRTC. Through WebRTC, I send audio to OME. Inside OME, I transcode the audio to AAC and push it via RTMP. The main idea to get HLS with audi track switching
Here is my config. I would appreciate your help.

?xml version="1.0" encoding="UTF-8" ?>

OvenMediaEngine origin * false stun.ovenmediaengine.com:13478

<Modules>
	<!-- 
	Currently OME only supports h2 like all browsers do. Therefore, HTTP/2 only works on TLS ports.			
	-->
	<HTTP2>
		<Enable>true</Enable>
	</HTTP2>

	<LLHLS>
		<Enable>true</Enable>
	</LLHLS>

	<!-- P2P works only in WebRTC and is experiment feature -->
	<P2P>
		<!-- disabled by default -->
		<Enable>false</Enable>
		<MaxClientPeersPerHostPeer>2</MaxClientPeersPerHostPeer>
	</P2P>
</Modules>

<!-- Settings for the ports to bind -->
<Bind>
	<!-- Enable this configuration if you want to use API Server -->
	<!--
	<Managers>
		<API>
			<Port>8081</Port>
			<TLSPort>8082</TLSPort>
			<WorkerCount>1</WorkerCount>
		</API>
	</Managers>
	-->
	<Providers>
		<!-- Pull providers -->
		<RTSPC>
			<WorkerCount>1</WorkerCount>
		</RTSPC>
		<OVT>
			<WorkerCount>1</WorkerCount>
		</OVT>
		<!-- Push providers -->
		<RTMP>
			<Port>${env:OME_RTMP_PROV_PORT:1935}</Port>
			<WorkerCount>1</WorkerCount>
		</RTMP>
		<SRT>
			<Port>${env:OME_SRT_PROV_PORT:9999}</Port>
			<WorkerCount>1</WorkerCount>
		</SRT>
		<!--
		<MPEGTS>
				<- Listen on port 4000
				This is just a demonstration to show that you can configure the port in several ways ->
				<Port>${env:OME_MPEGTS_PROV_PORT:4000-4001,4004,4005/udp}</Port>
			<Port>${env:OME_MPEGTS_PROV_PORT:4000-4005/udp}</Port>
		</MPEGTS>
		-->
		<WebRTC>
			<Signalling>
				<Port>${env:OME_WEBRTC_SIGNALLING_PORT:3333}</Port>
				<TLSPort>${env:OME_WEBRTC_SIGNALLING_TLS_PORT:3334}</TLSPort>
				<WorkerCount>1</WorkerCount>
			</Signalling>
			<IceCandidates>
				<IceCandidate>${env:OME_HOST_IP:*}:${env:OME_WEBRTC_CANDIDATE_PORT:10000-10004/udp}</IceCandidate>
				<TcpRelay>${env:OME_HOST_IP:*}:${env:OME_WEBRTC_TCP_RELAY_PORT:3478}</TcpRelay>
				<TcpForce>true</TcpForce>
				<TcpRelayWorkerCount>1</TcpRelayWorkerCount>
			</IceCandidates>
		</WebRTC>
	</Providers>

	<Publishers>
		<!-- The OVT is protocol for ORIGIN-EDGE -->
		<OVT>
			<Port>${env:OME_ORIGIN_PORT:9000}</Port>
			<WorkerCount>1</WorkerCount>
		</OVT>
		<LLHLS>
			<!-- 
			OME only supports h2, so LLHLS works over HTTP/1.1 on non-TLS port. 
			Note that LLHLS runs higher performance over HTTP/2.
			Therefore, it is recommended to use TLS Port.
			-->
			<Port>${env:OME_LLHLS_STREAM_PORT:3333}</Port>
			<TLSPort>${env:OME_LLHLS_STREAM_TLS_PORT:3334}</TLSPort>
			<WorkerCount>1</WorkerCount>
		</LLHLS>
		<WebRTC>
			<Signalling>
				<Port>${env:OME_WEBRTC_SIGNALLING_PORT:3333}</Port>
				<TLSPort>${env:OME_WEBRTC_SIGNALLING_TLS_PORT:3334}</TLSPort>
				<WorkerCount>1</WorkerCount>
			</Signalling>
			<IceCandidates>
				<IceCandidate>${env:OME_HOST_IP:*}:${env:OME_WEBRTC_CANDIDATE_PORT:10000-10004/udp}</IceCandidate>
				<TcpRelay>${env:OME_HOST_IP:*}:${env:OME_WEBRTC_TCP_RELAY_PORT:3478}</TcpRelay>
				<TcpForce>true</TcpForce>
				<TcpRelayWorkerCount>1</TcpRelayWorkerCount>
			</IceCandidates>
		</WebRTC>
	</Publishers>
</Bind>

<VirtualHosts>
	<!--
		You can include multiple XML files by doing the following:
		<VirtualHost include="sites-enabled/*.xml" />
	-->
	<VirtualHost include="VHost*.xml" />
	<VirtualHost>
		<Name>default</Name>
		<!--Distribution is a value that can be used when grouping the same vhost distributed across multiple servers. This value is output to the events log, so you can use it to aggregate statistics. -->
		<Distribution>ovenmediaengine.com</Distribution>
		
		<!-- Settings for multi ip/domain and TLS -->
		<Host>
			<Names>
				<!-- Host names
					<Name>stream1.airensoft.com</Name>
					<Name>stream2.airensoft.com</Name>
					<Name>*.sub.airensoft.com</Name>
					<Name>192.168.0.1</Name>
				-->
				<Name>*</Name>
			</Names>
			<TLS>
				<CertPath>./cert.crt</CertPath>
				<KeyPath>./cert.key</KeyPath>
				<ChainCertPath>./cert.ca-bundle</ChainCertPath>
			</TLS>
		</Host>

		<!-- Refer https://airensoft.gitbook.io/ovenmediaengine/signedpolicy
		<SignedPolicy>
			<PolicyQueryKeyName>policy</PolicyQueryKeyName>
			<SignatureQueryKeyName>signature</SignatureQueryKeyName>
			<SecretKey>aKq#1kj</SecretKey>

			<Enables>
				<Providers>rtmp,webrtc,srt</Providers>
				<Publishers>webrtc,llhls</Publishers>
			</Enables>
		</SignedPolicy>
		-->

		<!--
		<OriginMapStore>
			In order to use OriginMap, you must enable OVT Publisher in Origin and OVT Provider in Edge.
			<RedisServer>
				<Host>192.168.0.160:6379</Host>
				<Auth>!@#ovenmediaengine</Auth>
			</RedisServer>
			
			This is only needed for the origin server and used to register the ovt address of the stream. 
			<OriginHostName>${env:OME_HOST_IP:192.168.0.160}</OriginHostName>
		</OriginMapStore>
		-->

		<!-- Default CORS Settings -->
		<CrossDomains>
			<Url>*</Url>
		</CrossDomains>

		<!-- Settings for applications -->
		<Applications>
		<Application>
<Name>live</Name>
<Type>live</Type>
<Streams>
    <Stream>
        <Name>main_video_stream</Name>
        <Input>
            <WebRTC>
                <Url>wss://streamihive.live:3334/app/stream</Url>
            </WebRTC>
        </Input>
        <Outputs>
            <SelectOutputTrack>video</SelectOutputTrack>
        </Outputs>
    </Stream>

    <Stream>
        <Name>first_audio_stream</Name>
        <Input>
            <WebRTC>
                <Url>wss://streamihive.live:3334/app/stream</Url>
            </WebRTC>
        </Input>
        <Outputs>
            <Transcode>
                <Audio>
                    <Codec>aac</Codec>
                    <Bitrate>128000</Bitrate>
                    <Samplerate>44100</Samplerate>
                    <Channel>2</Channel>
                </Audio>
            </Transcode>
        </Outputs>
    </Stream>

    <Stream>
        <Name>second_audio_stream</Name>
        <Input>
            <WebRTC>
                <Url>wss://streamihive.live:3334/app/stream2</Url>
            </WebRTC>
        </Input>
        <Outputs>
            <Transcode>
                <Audio>
                    <Codec>aac</Codec>
                    <Bitrate>128000</Bitrate>
                    <Samplerate>44100</Samplerate>
                    <Channel>2</Channel>
                </Audio>
            </Transcode>
        </Outputs>
    </Stream>

    <Stream>
        <Name>mixed_stream</Name>
        <Input>
            <Stream>main_video_stream</Stream>
            <Stream>first_audio_stream</Stream>
            <Stream>second_audio_stream</Stream>
        </Input>
        <Outputs>
            <RTMP>
                <Url>rtmp://streamihive.live/live/mixed_stream</Url>
                <MultiAudio>
                    <Track1>first_audio_stream</Track1>
                    <Track2>second_audio_stream</Track2>
                </MultiAudio>
            </RTMP>
        </Outputs>
    </Stream>
</Streams>

			<Application>
				<Name>app</Name>
				<!-- Application type (live/vod) -->
				<Type>live</Type>
				<OutputProfiles>
					<!--  This option is deprecated. Please use the 'HWAccels' option. -->
					<!-- 
						<HardwareAcceleration>false</HardwareAcceleration>
					-->

					<!-- Enable this configuration if you want to hardware acceleration using GPU -->
					<HWAccels>
						<Decoder>
								<Enable>false</Enable>
								<!-- 
									Setting for Hardware Modules.
									 - xma :Xilinx Media Accelerator
									 - qsv :Intel Quick Sync Video
									 - nv : Nvidia Video Codec SDK
									 - nilogan: Netint VPU

									You can use multiple modules by separating them with commas.
									For example, if you want to use xma and nv, you can set it as follows.

									<Modules>[ModuleName]:[DeviceId],[ModuleName]:[DeviceId],...</Modules>
									<Modules>xma:0,nv:0</Modules>
								-->
								<!-- <Modules>nv</Modules> -->
						</Decoder>
						<Encoder>
								<Enable>false</Enable>
								<!-- <Modules>nv:0</Modules> -->
						</Encoder>
					</HWAccels>

					<OutputProfile>
						<Name>bypass_stream</Name>
						<OutputStreamName>${OriginStreamName}</OutputStreamName>

						<!-- 
						You can provide ABR with Playlist. Currently, ABR is supported in LLHLS and WebRTC.
						
						<Playlist>
							<Name>Paid</Name>
							You can play this playlist with 
							LLHLS : http[s]://<domain>[:port]/<app>/<stream>/<FileName>.m3u8 
							WebRTC : ws[s]://<domain>[:port]/<app>/<stream>/<FileName>
							Note that the keywords "playlist" and "chunklist" MUST NOT be included in FileName.
							<FileName>premium</FileName>

							Options is optional. 
							<Options>
								WebRtcAutoAbr : Default value is true
								<WebRtcAutoAbr>true</WebRtcAutoAbr> 
							</Options>

							<Rendition>
								<Name>bypass</Name>
								<Video>bypass_video</Video>
								<Audio>bypass_audio</Audio>
							</Rendition>
							<Rendition>
								<Name>480p</Name>
								<Video>480p</Video>
								<Audio>bypass_audio</Audio>
							</Rendition>
							<Rendition>
								<Name>720p</Name>
								<Video>720p</Video>
								<Audio>bypass_audio</Audio>
							</Rendition>
						</Playlist>
						<Playlist>
							<Name>free</Name>
							<FileName>free</FileName>
							<Rendition>
								<Name>720p</Name>
								<Video>720p</Video>
								<Audio>bypass_audio</Audio>
							</Rendition>
						</Playlist>
						-->

						<Encodes>
							<Video>
								<Name>bypass_video</Name>
								<Bypass>true</Bypass>
							</Video>
							<Audio>
								<Name>aac_audio</Name>
								<Codec>aac</Codec>
								<Bitrate>128000</Bitrate>
								<Samplerate>48000</Samplerate>
								<Channel>2</Channel>
								<BypassIfMatch>
									<Codec>eq</Codec>
								</BypassIfMatch>
							</Audio>								
							<Audio>
								<Name>opus_audio</Name>
								<Codec>opus</Codec>
								<Bitrate>128000</Bitrate>
								<Samplerate>48000</Samplerate>
								<Channel>2</Channel>
								<BypassIfMatch>
									<Codec>eq</Codec>
								</BypassIfMatch>									
							</Audio>
							<!-- 							
							<Video>
								<Name>video_1280</Name>
								<Codec>h264</Codec>
								<Bitrate>5024000</Bitrate>
								<Framerate>30</Framerate>
								<Width>1920</Width>
								<Height>1280</Height>
								<Preset>faster</Preset>
							</Video>
							<Video>
								<Name>video_720</Name>
								<Codec>h264</Codec>
								<Bitrate>2024000</Bitrate>
								<Framerate>30</Framerate>
								<Width>1280</Width>
								<Height>720</Height>
								<Preset>faster</Preset>
							</Video>
							-->
						</Encodes>
					</OutputProfile>
				</OutputProfiles>
				<Providers>
					<OVT />
					<WebRTC>
						<Timeout>30000</Timeout>
						<CrossDomains>
							<Url>*</Url>
						</CrossDomains>
					</WebRTC>
					<RTMP />
					<SRT />
					<RTSPPull />
					<Schedule>
						<MediaRootDir>/opt/ovenmediaengine/media</MediaRootDir>
						<ScheduleFilesDir>/opt/ovenmediaengine/media</ScheduleFilesDir>
					</Schedule>
					<!-- <MPEGTS>
						<StreamMap>
								<- Set the stream name of the client connected to the port to "stream_${Port}"
								For example, if a client connets to port 4000, OME creates a "stream_4000" stream ->
								<Stream>
									<Name>stream_${Port}</Name>
									<Port>4000,4001-4004</Port>
								</Stream>
								<Stream>
									<Name>stream_4005</Name>
									<Port>4005</Port>
								</Stream>
							<Stream>
								<Name>stream_${Port}</Name>
								<Port>${env:OME_MPEGTS_PROV_PORT:4000-4005/udp}</Port>
							</Stream>
						</StreamMap>
					</MPEGTS> -->
				</Providers>
				<Publishers>
					<AppWorkerCount>1</AppWorkerCount>
					<StreamWorkerCount>8</StreamWorkerCount>
					<OVT />
					<WebRTC>
						<Timeout>30000</Timeout>
						<Rtx>false</Rtx>
						<Ulpfec>false</Ulpfec>
						<JitterBuffer>false</JitterBuffer>
						<CreateDefaultPlaylist>true</CreateDefaultPlaylist>
					</WebRTC>
					<LLHLS>
						<OriginMode>true</OriginMode>
						<!--
						<CacheControl>
							<MasterPlaylistMaxAge>0</MasterPlaylistMaxAge>
							<ChunklistMaxAge>0</ChunklistMaxAge>
							<ChunklistWithDirectivesMaxAge>60</ChunklistWithDirectivesMaxAge>
							<SegmentMaxAge>-1</SegmentMaxAge>
							<PartialSegmentMaxAge>-1</PartialSegmentMaxAge>
						</CacheControl>
						-->
						<ChunkDuration>0.5</ChunkDuration>
						<!-- PartHoldBack SHOULD be at least three times the Part Target Duration -->
						<PartHoldBack>1.5</PartHoldBack>
						<SegmentDuration>6</SegmentDuration>
						<SegmentCount>10</SegmentCount>
						<CrossDomains>
							<Url>*</Url>
						</CrossDomains>
						<CreateDefaultPlaylist>true</CreateDefaultPlaylist> <!-- llhls.m3u8 -->
					</LLHLS>
					<HLS>
						<SegmentCount>4</SegmentCount>
						<SegmentDuration>4</SegmentDuration>
						<!-- <DVR>
							<Enable>true</Enable>
							<EventPlaylistType>false</EventPlaylistType>
							<TempStoragePath>/tmp/ome_dvr/</TempStoragePath>
							<MaxDuration>600</MaxDuration>
						</DVR> -->
						<CrossDomains>
							<Url>*</Url>
						</CrossDomains>
						<CreateDefaultPlaylist>true</CreateDefaultPlaylist> <!-- ts:playlist.m3u8 -->
					</HLS>
				</Publishers>
			</Application>
		</Applications>
	</VirtualHost>
</VirtualHosts>