Solution for commentators. Ingest RTMP and display via WEBRTC. Comment via WebRTC and mix it to hls #1743

VitalyZaikin · 2024-11-28T17:32:29Z

VitalyZaikin
Nov 28, 2024

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>

Answered by getroot

Dec 2, 2024

Mux Channel is a feature that allows you to take a stream created inside OME and reassemble the tracks to create a new channel. You cannot enter a URL starting with wss://, and according to the manual, enter information about a stream already created with WebRTC according to the stream://vhost_name/app_name/stream_name rule.

View full answer

nums · 2024-11-29T16:54:16Z

nums
Nov 29, 2024

You can take a look here : https://airensoft.gitbook.io/ovenmediaengine/live-source/multiplex-channel

5 replies

VitalyZaikin Dec 1, 2024
Author

@nums Thanks for the answer. It still doesn't work for me, don't know why. Could you share some examples of worked config?

Here is my Server config

30000 * /opt/ovenmediaengine/media /opt/ovenmediaengine/media /usr/share/ovenmediaengine/conf/

And here is mux.mux file

mux

<SourceStreams>
    <SourceStream>
        <Name>tv1</Name>
        <Url>wss://streamihive.live:3334/app/stream2</Url>
        <TrackMap>
            <Track>
                <SourceTrackName>bypass_video</SourceTrackName>
                <NewTrackName>tv1_video</NewTrackName>
            </Track>
            <Track>
                <SourceTrackName>bypass_audio</SourceTrackName>
                <NewTrackName>tv1_audio</NewTrackName>
            </Track>
            <Track>
                <SourceTrackName>opus</SourceTrackName>
                <NewTrackName>tv1_opus</NewTrackName>
            </Track>
        </TrackMap>
    </SourceStream>
    <SourceStream>
        <Name>tv2</Name>
        <Url>wss://streamihive.live:3334/app/stream2</Url>

        <TrackMap>
            <Track>
                <SourceTrackName>bypass_video</SourceTrackName>
                <NewTrackName>tv2_video</NewTrackName>
            </Track>
            <Track>
                <SourceTrackName>bypass_audio</SourceTrackName>
                <NewTrackName>tv2_audio</NewTrackName>
            </Track>
            <Track>
                <SourceTrackName>opus</SourceTrackName>
                <NewTrackName>tv2_opus</NewTrackName>
            </Track>
        </TrackMap>

    </SourceStream>
</SourceStreams>

<Playlists>
    <Playlist>
        <Name>123</Name>
        <FileName>abr</FileName>
        
        <Rendition>
            <Name>1080p</Name>
            <Video>tv1_video</Video>
            <Audio>tv1_audio</Audio>
        </Rendition>
        <Rendition>
            <Name>720p</Name>
            <Video>tv2_video</Video>
            <Audio>tv2_audio</Audio>
        </Rendition>
    </Playlist>
</Playlists>

I really can't recognize where are the issues

getroot Dec 2, 2024
Maintainer Sponsor

Mux Channel is a feature that allows you to take a stream created inside OME and reassemble the tracks to create a new channel. You cannot enter a URL starting with wss://, and according to the manual, enter information about a stream already created with WebRTC according to the stream://vhost_name/app_name/stream_name rule.

Answer selected by VitalyZaikin

VitalyZaikin Dec 3, 2024
Author

Thanks, was a misunderstanding about the stream format. Just for understanding, is it possible to mix audio pairs, like mono, stereo?

nums Dec 3, 2024

For now, there is not mixing feature or channel audio track management but you can provide 2 HLS streams : one with video studio + audio studio and the other with video studio + audio commentators

VitalyZaikin Dec 10, 2024
Author

For now, there is not mixing feature or channel audio track management but you can provide 2 HLS streams : one with video studio + audio studio and the other with video studio + audio commentators

Thanks, really helpful. One more question, does webtrtc stream streaming in stereo, or it's mono in 2 pairs?

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Solution for commentators. Ingest RTMP and display via WEBRTC. Comment via WebRTC and mix it to hls #1743

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Solution for commentators. Ingest RTMP and display via WEBRTC. Comment via WebRTC and mix it to hls #1743

VitalyZaikin Nov 28, 2024

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