The VirtualHost is read-only: [default] (403)

[Nikita-Goncharov]

I Have that error in postman:

{
    "message": "[HTTP] The VirtualHost is read-only: [default] (403)",
    "statusCode": 403
}

When i try to create new application, create or delete outputprofile.
But getting vhosts, apps and recording streams in files works.

Example of output profile creation:

Link: myIP:3005/v1/vhosts/default/apps/app/outputProfiles
Right Authorization token in headers.
Body:

[
  {
    "name": "bypass_profile",
    "outputStreamName": "stream_bypass",
    "encodes": {
      "videos": [
        {
          "bypass": true
        }
      ],
      "audios": [
        {
          "bypass": true
        }
      ]
    }
  }
]

OvenMediaEngine Server.xml config:

<?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>3005</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>
        </Host>
        <API>
            <AccessToken>myUser:myPassword</AccessToken>

            <CrossDomains>
                <Url>*</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>
                        <FILE>  
                            <!-- [Optional] -->
                            <RootPath>/mnt/shared_volumes</RootPath>
                            
                            <!-- [Must] Recorded file and info path 
                                When recording starts via the API, the following path is used as 
                                the default path. If a path is set via the API, it will not be used.
                            -->
                            <FilePath>/${VirtualHost}/${Application}/${Stream}/
                                ${StartTime:YYYYMMDDhhmmss}_${EndTime:YYYYMMDDhhmmss}.ts</FilePath>
                            <InfoPath>/${VirtualHost}/${Application}/${Stream}.xml</InfoPath>
                            
                            <!-- [Optional] Recording settings for file-based automatic recording -->
                            <Record>
                                <Enable>true</Enable>
                                <RecordInfo>./record_info.xml</RecordInfo>
                            </Record>
                        </FILE>

                        <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>

What should i do ?🙏

[getroot]

This is by design and the Virtual Host created from Server.xml is read only. Only virtual hosts created through the API can be modified.

[Nikita-Goncharov]

You really helped to me, but i need your help again🙏

I have a problem with sharing stream on custom virtualhost.
I created virtual host using API, there is data:

[
    {
        "name": "vhost",
        "host": {
            "names": [
                "*"
            ]
        },

        "signedPolicy": {
            "enables": {
                "providers": "rtmp,webrtc,srt",
                "publishers": "webrtc,llhls"
            },
            "policyQueryKeyName": "policy",
            "secretKey": "aKq#1kj",
            "signatureQueryKeyName": "signature"
        },
        "origins": {
            "origin": [
                {
                    "location": "/custom_app/stream",
                    "pass": {
                        "scheme": "rtsp",
                        "urls": {
                            "url": [
                                "rtsp.server:8554/ca-01"
                            ]
                        }
                    }
                }
            ]
        }
    }
]

Then I create app on this host(in url i add "vhost", not "default"):

[
    {
        "name": "custom_app",
        "type": "live",
        "outputProfiles": {
            "outputProfile": [
                {
                    "name": "stream",
                    "outputStreamName": "stream",
                    "encodes": {
                        "audios": [
                            {
                                "name": "opus",
                                "codec": "opus",
                                "samplerate": 48000,
                                "bitrate": 128000,
                                "channel": 2,
                                "bypassIfMatch": {
                                    "codec": "eq"
                                }
                            },
                            {
                                "name": "aac",
                                "codec": "aac",
                                "samplerate": 48000,
                                "bitrate": 128000,
                                "channel": 2,
                                "bypassIfMatch": {
                                    "codec": "eq"
                                }
                            }
                        ],
                        "videos": [
                            {
                                "name": "bypass_video",
                                "bypass": true
                            }
                        ]
                    }
                }
            ]
        },
        "providers": {
            "ovt": {},
            "rtmp": {},
            "rtspPull": {},
            "srt": {},
            "webrtc": {}
        },
        "publishers": {
            "llhls": {},
            "ovt": {},
            "webrtc": {}
        }
    }
]

For testing i use OBS studio and when i try pass stream:
rtmp://192.168.0.106:1935/custom_app
secret key: stream

In docker container i receive that log:
2024-05-20 13:14:56 [2024-05-20 10:14:56.486] E [SPRTMP-t1935:28] RTMPProvider | rtmp_stream.cpp:442 | Could not find application: #default#custom_app
2024-05-20 13:14:56 [2024-05-20 10:14:56.486] E [SPRTMP-t1935:28] RTMPProvider | rtmp_stream.cpp:341 | SendAmfCreateStreamResult Fail

I understand that the server is trying to find custom_app on the default vhost, but I don't know what to do, maybe I have set the wrong configuration for vhost/app?
Please, help me.

[Nikita-Goncharov]

And another general question: I need to organize the server in such a way that I can add new streams from cameras, links to streams from which I will need to change frequently for security reasons. I'm going to create new applications when I add a new camera and after a while delete the outputProfiles, and then create new ones with a different outputStreamName. And as I understand, after such actions there is no need to restart the server, right? Or maybe you can recommend another scheme of work? You would help me a lot with this

[getroot]

https://airensoft.gitbook.io/ovenmediaengine/configuration#host

The existing Virtual Host will process your request first. Delete the existing Virtual Host or change the Host rules.

[getroot]

And we ask you to stop putting pressure on us by continuously sending emails or posting repeatedly on the community.

[getroot]

If you want to discard distributed URLs, use the Access Control function. Adding and deleting OutputProfiles for that purpose are not used for that purpose.

https://airensoft.gitbook.io/ovenmediaengine/access-control