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AirenSoft / OvenMediaEngine

OvenMediaEngine (OME) is a Sub-Second Latency Live Streaming Server with Large-Scale and High-Definition. #WebRTC #LLHLS
https://airensoft.com/ome.html
GNU Affero General Public License v3.0
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Question: How to verify OME is receiving SRT data #1659

Closed traviszuleger closed 4 months ago

traviszuleger commented 4 months ago

I want to preface that I have never used OME before so this could be a very dumb question, and I may have set it up wrong too. I am using the default configuration. I will post the Server.xml file below as well as the Docker run command I used to start OME.

I'm trying to stream AAC/H264 samples from a Node.js app to OME. My goal is to get a Low Latency HLS stream out of it. The best way I could figure it out was by manually creating the TS packets myself and then piping that data into FFmpeg which then sends it to the SRT port.

I keep my OME demo server as Docker Container on a remote ubuntu 20.04 machine I access over my VPN. My program doesn't crash and it doesn't give me any errors, but I have no way of knowing that OME is actually getting the live stream.

I've tried navigating to https://{ip}:3334/app/test/llhls.m3u8 but I don't think I have my certs set up right (I use the Docker container jwilder/nginx-proxy with the Docker container nginxproxy/acme-companion) so I tried navigating to http://{ip}:3334/app/test/ts:playlist.m3u8 instead.

Here is my Server.xml file (same for both edge and origin conf):

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

<Server version="8">
    <Name>OvenMediaEngine</Name>
    <!-- Host type (origin/edge) -->
    <Type>origin</Type>
    <!-- Specify IP address to bind (* means all IPs) -->
    <IP>*</IP>
    <PrivacyProtection>false</PrivacyProtection>

    <!--
    To get the public IP address(mapped address of stun) of the local server.
    This is useful when OME cannot obtain a public IP from an interface, such as AWS or docker environment.
    If this is successful, you can use ${PublicIP} in your settings.
    -->
    <StunServer>stun.l.google.com:19302</StunServer>

    <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>1935</Port>
            <WorkerCount>1</WorkerCount>
        </RTMP>
        <SRT>
            <Port>9999</Port>
            <WorkerCount>1</WorkerCount>
        </SRT>
        <MPEGTS>
            <!--
                Listen on port 4000~4005 (<Port>4000-4004,4005/udp</Port>)
                This is just a demonstration to show that you can configure the port in several ways
            -->
            <Port>4000/udp</Port>
        </MPEGTS>
        <WebRTC>
            <Signalling>
                <Port>3333</Port>
                <TLSPort>3334</TLSPort>
                <WorkerCount>1</WorkerCount>
            </Signalling>

            <IceCandidates>
                <IceCandidate>*:10000/udp</IceCandidate>
                <!--
                    If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
                    This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP.
                    For detailed information, refer https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#webrtc-over-tcp
                -->
                <TcpRelay>*:3478</TcpRelay>
                <!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
                <TcpForce>true</TcpForce>
                <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
        </WebRTC>
    </Providers>

    <Publishers>
        <OVT>
            <Port>9000</Port>
            <WorkerCount>1</WorkerCount>
        </OVT>
        <LLHLS>
            <!--
            OME only supports h2, so LLHLS works over HTTP/1.1 on non-TLS ports.
            LLHLS works with higher performance over HTTP/2,
            so it is recommended to use a TLS port.
            -->
            <Port>3333</Port>
            <!-- If you want to use TLS, specify the TLS port -->
            <TLSPort>3334</TLSPort>
            <WorkerCount>1</WorkerCount>
        </LLHLS>
        <WebRTC>
            <Signalling>
                <Port>3333</Port>
                <TLSPort>3334</TLSPort>
                <WorkerCount>1</WorkerCount>
            </Signalling>
            <IceCandidates>
                <IceCandidate>*:10000-10005/udp</IceCandidate>
                <!--
                    If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
                    This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP.
                    For detailed information, refer https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#webrtc-over-tcp
                -->
                <TcpRelay>*:3478</TcpRelay>
                <!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
                <TcpForce>true</TcpForce>
                <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
        </WebRTC>
    </Publishers>
</Bind>

    <!--
        Enable this configuration if you want to use API Server

        <AccessToken> is a token for authentication, and when you invoke the API, you must put "Basic base64encode(<AccessToken>)" in the "Authorization" header of HTTP request.
        For example, if you set <AccessToken> to "ome-access-token", you must set "Basic b21lLWFjY2Vzcy10b2tlbg==" in the "Authorization" header.
    -->
    <!--
    <Managers>
        <Host>
            <Names>
                <Name>*</Name>
            </Names>
            <TLS>
                <CertPath>path/to/file.crt</CertPath>
                <KeyPath>path/to/file.key</KeyPath>
                <ChainCertPath>path/to/file.crt</ChainCertPath>
            </TLS>
        </Host>
        <API>
            <AccessToken>ome-access-token</AccessToken>

            <CrossDomains>
                <Url>*.airensoft.com</Url>
                <Url>http://*.sub-domain.airensoft.com</Url>
                <Url>http?://airensoft.*</Url>
            </CrossDomains>
        </API>
    </Managers>
    -->

    <VirtualHosts>
        <!-- You can use wildcard like this to include multiple XMLs -->
        <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>path/to/file.crt</CertPath>
                    <KeyPath>path/to/file.key</KeyPath>
                    <ChainCertPath>path/to/file.crt</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,hls,llhls,dash,lldash</Publishers>
                </Enables>
            </SignedPolicy>
            -->

            <!--
            <AdmissionWebhooks>
                <ControlServerUrl></ControlServerUrl>
                <SecretKey></SecretKey>
                <Timeout>3000</Timeout>
                <Enables>
                    <Providers>rtmp,webrtc,srt</Providers>
                    <Publishers>webrtc,hls,llhls,dash,lldash</Publishers>
                </Enables>
            </AdmissionWebhooks>
            -->

            <!-- <Origins>
                <Properties>
                    <NoInputFailoverTimeout>3000</NoInputFailoverTimeout>
                    <UnusedStreamDeletionTimeout>60000</UnusedStreamDeletionTimeout>
                </Properties>
                <Origin>
                    <Location>/app/stream</Location>
                    <Pass>
                        <Scheme>ovt</Scheme>
                        <Urls><Url>origin.com:9000/app/stream_720p</Url></Urls>
                    </Pass>
                    <ForwardQueryParams>false</ForwardQueryParams>
                </Origin>
                <Origin>
                    <Location>/app/</Location>
                    <Pass>
                        <Scheme>ovt</Scheme>
                        <Urls><Url>origin.com:9000/app/</Url></Urls>
                    </Pass>
                </Origin>
                <Origin>
                    <Location>/edge/</Location>
                    <Pass>
                        <Scheme>ovt</Scheme>
                        <Urls><Url>origin.com:9000/app/</Url></Urls>
                    </Pass>
                </Origin>
            </Origins> -->

            <!-- Settings for applications -->
            <Applications>
                <Application>
                    <Name>app</Name>
                    <!-- Application type (live/vod) -->
                    <Type>live</Type>
                    <OutputProfiles>
                        <!-- Enable this configuration if you want to hardware acceleration using GPU -->
                        <HardwareAcceleration>false</HardwareAcceleration>
                        <OutputProfile>
                            <Name>bypass_stream</Name>
                            <OutputStreamName>${OriginStreamName}</OutputStreamName>
                            <Encodes>
                                <Audio>
                                    <Bypass>true</Bypass>
                                </Audio>
                                <Video>
                                    <Bypass>true</Bypass>
                                </Video>
                                <Audio>
                                    <Codec>opus</Codec>
                                    <Bitrate>128000</Bitrate>
                                    <Samplerate>48000</Samplerate>
                                    <Channel>2</Channel>
                                </Audio>
                                <!--
                                <Video>
                                    <Codec>vp8</Codec>
                                    <Bitrate>1024000</Bitrate>
                                    <Framerate>30</Framerate>
                                    <Width>1280</Width>
                                    <Height>720</Height>
                                    <Preset>faster</Preset>
                                </Video>
                                -->
                            </Encodes>
                        </OutputProfile>
                    </OutputProfiles>
                    <Providers>
                        <OVT />
                        <WebRTC />
                        <RTMP />
                        <SRT />
                        <MPEGTS>
                            <StreamMap>
                                <!--
                                    Set the stream name of the client connected to the port to "stream_${Port}"
                                    For example, if a client connects 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>4000</Port>
                                </Stream>
                            </StreamMap>
                        </MPEGTS>
                        <RTSPPull />
                        <WebRTC>
                            <Timeout>30000</Timeout>
                        </WebRTC>
                    </Providers>
                    <Publishers>
                        <AppWorkerCount>1</AppWorkerCount>
                        <StreamWorkerCount>8</StreamWorkerCount>
                        <OVT />
                        <WebRTC>
                            <Timeout>30000</Timeout>
                            <Rtx>false</Rtx>
                            <Ulpfec>false</Ulpfec>
                            <JitterBuffer>false</JitterBuffer>
                        </WebRTC>
                        <LLHLS>
                            <ChunkDuration>0.2</ChunkDuration>
                            <SegmentDuration>6</SegmentDuration>
                            <SegmentCount>10</SegmentCount>
                            <CrossDomains>
                                <Url>*</Url>
                            </CrossDomains>
                        </LLHLS>
                    </Publishers>
                </Application>
            </Applications>
        </VirtualHost>
    </VirtualHosts>
</Server>

Here is my docker run command (shell script):

sudo docker rm -f oven-media-engine

HOST_IP=$(curl ipv4.icanhazip.com)
docker run -d --name oven-media-engine \
        -v ~/OvenMediaEngine/mounts/logs/:/var/log/ovenmediaengine/ \
        -v ~/OvenMediaEngine/mounts/ome-origin-conf/:/opt/ovenmediaengine/bin/origin_conf/ \
        -v ~/OvenMediaEngine/mounts/ome-edge-conf/:/opt/ovenmediaengine/bin/edge_conf/ \
        -v ~/nginx/sslconf/certs/:/cert/ \
        --network nginx-net \
        -e LETSENCRYPT_HOST=ome.{mydomain}.com,www.ome.{mydomain}.com \
        -e VIRTUAL_PORT=3334 \
        -e OME_HOST_IP=$HOST_IP \
        --expose 3333 \
        --expose 3334 \
        --expose 9999 \
        -p 1935:1935 \
        -p 9999:9999 \
        -p 9999:9999/udp \
        -p 9000:9000 \
        -p 3333:3333 \
        -p 3334:3334 \
        -p 3478:3478 \
        -p 10000-10009:10000-10009/udp \
        airensoft/ovenmediaengine:latest

Here is my ffmpeg command (spawned and stdin piped from Node.js):

ffmpeg -re -stream_loop -i - -c copy srt://192.168.1.100:9999?streamid=srt%3A%2F%2F192.168.1.100%3A9999%2Fapp%2Ftest&latency=2000000

I would appreciate any help. I've been down the depths of live streaming for 6 months now and I think this might be the cleanest solution I could make to achieve LL-HLS from raw audio/video samples. (Every piece of documentation uses a file like .mp4 as input or RTSP server, etc. I require a solution that works explicitly on just raw AAC/H264 audio video samples.)

I also apologize in advance if this question should be posted elsewhere.

traviszuleger commented 4 months ago

Posting to the help discussion page, as I think this is better off there as to not flood the issue section. My apologies.

https://github.com/AirenSoft/OvenMediaEngine/discussions/1660