OME worked before, but can't get it working behind double NAT with Tunnel and VPN to route traffic #1238

devedse · 2023-05-17T09:44:30Z

devedse
May 17, 2023

Introduction
A while back I setup "OpenStreamingPlatform (OSP)" to host some small game streams for me and my friends. A while after that OSP added the functionality to also integrate with "Oven Media Engine (OME)". It took me a bit to setup but after a while that also worked great.

As far as I understand I've been using TCPRelay then since that was in the default config. I only changed some minor things.

Here's the most relevant parts of my configuration (I removed some of the last parts in here as I don't think that part is relevant):

Variables:

MYOLDIP: Would be the old IP address of my home

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

<Server version="8">
        <Name>OvenMediaEngine</Name>
        <Type>origin</Type>
        <IP>*</IP>
        <PrivacyProtection>false</PrivacyProtection>
        <StunServer>stun.ovenmediaengine.com:13478</StunServer>

        <Modules>
                <HTTP2>
                        <Enable>true</Enable>
                </HTTP2>
                <LLHLS>
                        <Enable>true</Enable>
                </LLHLS>
                <P2P>
                        <Enable>false</Enable>
                        <MaxClientPeersPerHostPeer>2</MaxClientPeersPerHostPeer>
                </P2P>
        </Modules>

        <Bind>
                <Providers>
                        <RTSPC>
                                <WorkerCount>1</WorkerCount>
                        </RTSPC>
                        <OVT>
                                <WorkerCount>1</WorkerCount>
                        </OVT>
                        <RTMP>
                                <Port>1935</Port>
                                <WorkerCount>1</WorkerCount>
                        </RTMP>
                        <SRT>
                                <Port>9999</Port>
                                <WorkerCount>1</WorkerCount>
                        </SRT>
                        <MPEGTS>
                                <Port>4000/udp</Port>
                        </MPEGTS>
                        <WebRTC>
                                <Signalling>
                                        <Port>3333</Port>
                                        <TLSPort>3334</TLSPort>
                                        <WorkerCount>2</WorkerCount>
                                </Signalling>
                                <IceCandidates>
                                        <IceCandidate>*:10000-10005/udp</IceCandidate>
                                        <TcpRelay>*:3478</TcpRelay>
                                        <TcpForce>false</TcpForce>
                                        <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
                                </IceCandidates>
                        </WebRTC>
                </Providers>

                <Publishers>
                        <OVT>
                                <Port>9000</Port>
                                <WorkerCount>1</WorkerCount>
                        </OVT>
                        <LLHLS>
                                <Port>3333</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>
                                        <TcpRelay>*:3478</TcpRelay>
                                        <TcpForce>true</TcpForce>
                                        <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
                                </IceCandidates>
                        </WebRTC>
                </Publishers>
        </Bind>
        <VirtualHosts>
                <VirtualHost include="VHost*.xml" />
                <VirtualHost>
                        <Name>default</Name>
                        <Distribution>********.duckdns.org</Distribution>
                        <Host>
                                <Names>
                                        <Name>********.duckdns.org</Name>
                                        <Name>{MYOLDIP}</Name>
                                        <Name>*</Name>
                                </Names>
                                <TLS>
                                        <CertPath>/etc/letsencrypt/live/********.duckdns.org/cert.pem</CertPath>
                                        <KeyPath>/etc/letsencrypt/live/********.duckdns.org/privkey.pem</KeyPath>
                                        <ChainCertPath>/etc/letsencrypt/live/********.duckdns.org/fullchain.pem</ChainCertPath>
                                </TLS>
                        </Host>
                </VirtualHost>
        </VirtualHosts>
</Server>

Describe the bug

Recently I moved to a new place where I don't have access to an internet provider that can provide me with a public IP address at all. So I'm basically behind double NAT. A friend of mine bought multiple IP Addresses though so we decided to setup a Tunnel and route all traffic through that IP Address to me through that Tunnel. It was a bit of a hassle to setup but also after a bit I managed to make OSP working again.

I then started to try to get OME working again as well but I keep running into issues.

Since I still don't quite understand what ports I do need to forward and which ones are internal I decided to forward the following ports:

80/TCP (For OSP)
443/TCP (For OSP)
1935/TCP (For RTMP)
3333/TCP
3334/TCP
3478/TCP
3478/UDP
9000/TCP
9999/TCP
10000-10005/UDP

I keep running into multiple issues with this setup:

1. Could not parse STUN packet

osp_ovenmediaengine       | [2023-05-17 09:17:05.655] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>
osp_ovenmediaengine       | [2023-05-17 09:17:16.155] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>
osp_ovenmediaengine       | [2023-05-17 09:17:16.156] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>

When I check my docker logs I see this message all over the place. I'm sure this wasn't the case before.

From what I understand a STUN server helps determine the public IP Address of OME. Since I'm behind a double NAT and we should actually be using my friends IP Address I changed the following config sections:

<Providers>
    ....
    <WebRTC>
            <Signalling>
                    <Port>3333</Port>
                    <TLSPort>3334</TLSPort>
                    <WorkerCount>2</WorkerCount>
            </Signalling>
            <IceCandidates>
                    <IceCandidate>{FRIENDIP}:10000-10005/udp</IceCandidate>    <--- Changed this
                    <TcpRelay>{FRIENDIP}:3478</TcpRelay>    <--- Changed this
                    <TcpForce>false</TcpForce>
                    <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
    </WebRTC>
</Providers>

<Publishers>
    ....
    <WebRTC>
            <Signalling>
                    <Port>3333</Port>
                    <TLSPort>3334</TLSPort>
                    <WorkerCount>2</WorkerCount>
            </Signalling>
            <IceCandidates>
                    <IceCandidate>{FRIENDIP}:10000-10005/udp</IceCandidate>    <--- Changed this
                    <TcpRelay>{FRIENDIP}:3478</TcpRelay>   <--- Changed this
                    <TcpForce>false</TcpForce>    <--- Changed this
                    <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
    </WebRTC>
</Publishers>

And I did not change:

<IP>*</IP>
<PrivacyProtection>false</PrivacyProtection>
<StunServer>stun.ovenmediaengine.com:13478</StunServer>

I have been playing around with TcpForce false and true but haven't been able to make it work in either scenario.

2. wss connection in browser doesn't seem to connect

When I start streaming the logs seem to show correcty that OME has accepted the stream and started the WebRTC publisher.

If I go to the streaming page then, I see the following logs in my Chrome Dev Tools:

And this is what the logs show on Docker:

To Reproduce
I'm using the latest version of both OSP and OME. But it's somewhat hard to give an easy reproduction scenario.

Expected behavior
When I open my stream I would expect the WebRTC stream to play and not fall back to the OSP provided HLS stream. (And skip OME alltogether)

Logs

osp_ovenmediaengine       | [2023-05-17 09:29:42.136] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>
osp_ovenmediaengine       | [2023-05-17 09:29:49.421] I [SPRtcSig-t3334:13] HttpServer | http_server.cpp:149  | Client(<ClientSocket: 0x7f7c447e10, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:59026>) is connected on *:3334
osp_ovenmediaengine       | [2023-05-17 09:29:49.500] I [SPRtcSig-t3334:13] HttpServer | http_connection.cpp:147  | TLS connection accepted : Server Name(*************.duckdns.org) Alpn Protocol(http/1.1) Client (<ClientSocket: 0x7f7c447e10, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:59026>)
osp_ovenmediaengine       | [2023-05-17 09:29:49.504] I [SPRtcSig-t3334:13] Signalling | rtc_signalling_server.cpp:316  | New client is connected: <ClientSocket: 0x7f7c447e10, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:59026>
osp_ovenmediaengine       | [2023-05-17 09:29:49.566] I [SPRtcSig-t3334:13] Signalling | rtc_signalling_server.cpp:450  | Client is disconnected: HttpConnection(0x7f88001810) : WebSocket <ClientSocket: 0x7f7c447e10, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:59026> TLS(Enabled) (#default#app / d406***********************79e16, ufrag: local: 2cUkn1, remote: (N/A))
osp_ovenmediaengine       | [2023-05-17 09:29:49.568] I [SPRtcSig-t3334:13] HttpServer | http_server.cpp:216  | Client(<ClientSocket: 0x7f7c447e10, #33, Closed, TCP, Nonblocking, ***{internalOMEIP}***:59026>) has been disconnected by *:3334
osp_ovenmediaengine       | [2023-05-17 09:29:49.618] I [SPRtcSig-t3334:13] HttpServer | http_server.cpp:149  | Client(<ClientSocket: 0x7f7c44a610, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:59034>) is connected on *:3334
osp_ovenmediaengine       | [2023-05-17 09:29:49.664] I [SPRtcSig-t3334:13] HttpServer | http_connection.cpp:147  | TLS connection accepted : Server Name(*************.duckdns.org) Alpn Protocol(http/1.1) Client (<ClientSocket: 0x7f7c44a610, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:59034>)
osp_ovenmediaengine       | [2023-05-17 09:29:49.674] I [SPRtcSig-t3334:13] Signalling | rtc_signalling_server.cpp:316  | New client is connected: <ClientSocket: 0x7f7c44a610, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:59034>
osp_ovenmediaengine       | [2023-05-17 09:29:49.803] I [SPRtcSig-t3334:13] Monitor | stream_metrics.cpp:114  | A new session has started playing #default#app/d406***********************79e16 on the WebRTC publisher. WebRTC(1)/Stream total(1)/App total(1)
osp_ovenmediaengine       | [2023-05-17 09:29:49.803] I [SPRtcSig-t3334:13] ICE | ice_port.cpp:317  | Added session: 116 (ufrag: 5COGX0:xH+K)
osp_ovenmediaengine       | [2023-05-17 09:29:49.811] I [SPICE-u10001:19] ICE | ice_port.cpp:868  | Add the client to the port list: ***{internalOMEIP}***:34871 / Packet type : STUN GateType : DIRECT
osp_ovenmediaengine       | [2023-05-17 09:31:37.874] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>
osp_ovenmediaengine       | [2023-05-17 09:31:38.256] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>
...
osp_ovenmediaengine       | [2023-05-17 09:36:32.086] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>
osp_ovenmediaengine       | [2023-05-17 09:36:32.087] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>
osp_ovenmediaengine       | [2023-05-17 09:36:32.914] I [SPRtcSig-t3334:13] WebRTC Publisher | webrtc_publisher.cpp:547  | Stop command received : #default#app/d406***********************79e16/116
osp_ovenmediaengine       | [2023-05-17 09:36:32.916] I [SPRtcSig-t3334:13] Monitor | stream_metrics.cpp:136  | A session has been stopped playing #default#app/d406***********************79e16 on the WebRTC publisher. Concurrent Viewers[WebRTC(0)/Stream total(0)/App total(0)]
osp_ovenmediaengine       | [2023-05-17 09:36:32.917] I [SPRtcSig-t3334:13] Signalling | rtc_signalling_server.cpp:450  | Client is disconnected: HttpConnection(0x7f7c447e10) : WebSocket <ClientSocket: 0x7f7c44a610, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:59034> TLS(Enabled) (#default#app / d406***********************79e16, ufrag: local: 5COGX0, remote: xH+K)
osp_ovenmediaengine       | [2023-05-17 09:36:32.918] I [SPRtcSig-t3334:13] HttpServer | http_server.cpp:216  | Client(<ClientSocket: 0x7f7c44a610, #33, Closed, TCP, Nonblocking, ***{internalOMEIP}***:59034>) has been disconnected by *:3334
osp_ovenmediaengine       | [2023-05-17 09:36:33.799] I [SPRtcSig-t3334:13] HttpServer | http_server.cpp:149  | Client(<ClientSocket: 0x7f88001410, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:37774>) is connected on *:3334
osp_ovenmediaengine       | [2023-05-17 09:36:33.869] I [SPRtcSig-t3334:13] HttpServer | http_connection.cpp:147  | TLS connection accepted : Server Name(*************.duckdns.org) Alpn Protocol(http/1.1) Client (<ClientSocket: 0x7f88001410, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:37774>)
osp_ovenmediaengine       | [2023-05-17 09:36:33.878] I [SPRtcSig-t3334:13] Signalling | rtc_signalling_server.cpp:316  | New client is connected: <ClientSocket: 0x7f88001410, #33, Connected, TCP, Nonblocking, ***{internalOMEIP}***:37774>
osp_ovenmediaengine       | [2023-05-17 09:36:33.970] I [SPRtcSig-t3334:13] HttpServer | http_server.cpp:149  | Client(<ClientSocket: 0x7f7c447e10, #34, Connected, TCP, Nonblocking, ***{internalOMEIP}***:37776>) is connected on *:3334
osp_ovenmediaengine       | [2023-05-17 09:36:34.018] I [SPRtcSig-t3334:13] HttpServer | http_connection.cpp:147  | TLS connection accepted : Server Name(*************.duckdns.org) Alpn Protocol(http/1.1) Client (<ClientSocket: 0x7f7c447e10, #34, Connected, TCP, Nonblocking, ***{internalOMEIP}***:37776>)
osp_ovenmediaengine       | [2023-05-17 09:36:34.033] I [SPRtcSig-t3334:13] Signalling | rtc_signalling_server.cpp:316  | New client is connected: <ClientSocket: 0x7f7c447e10, #34, Connected, TCP, Nonblocking, ***{internalOMEIP}***:37776>
osp_ovenmediaengine       | [2023-05-17 09:36:34.174] I [SPRtcSig-t3334:13] Monitor | stream_metrics.cpp:114  | A new session has started playing #default#app/d406***********************79e16 on the WebRTC publisher. WebRTC(1)/Stream total(1)/App total(1)
osp_ovenmediaengine       | [2023-05-17 09:36:34.175] I [SPRtcSig-t3334:13] ICE | ice_port.cpp:317  | Added session: 117 (ufrag: jkdH3a:ddJ6)
osp_ovenmediaengine       | [2023-05-17 09:36:34.217] I [SPICE-u10000:20] ICE | ice_port.cpp:868  | Add the client to the port list: ***{internalOMEIP}***:34108 / Packet type : STUN GateType : DIRECT
osp_ovenmediaengine       | [2023-05-17 09:36:38.260] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>
osp_ovenmediaengine       | [2023-05-17 09:36:38.261] E [SPICE-u10001:19] ICE | ice_port.cpp:720  | Could not parse STUN packet from <DatagramSocket: 0x7f8cbece10, #19, Bound, UDP, Nonblocking>

Server (please complete the following information):

OS: Docker Linux ARM64 (It did work previously)
OvenMediaEngine Version: Latest Docker Master as of 17/05/2023
Branch: N/A

Player (please complete the following information):

OS: Windows 11
Browser: Chrome
Version: 113.0.5672.93

Additional context
N/A

Answered by devedse

May 19, 2023

Ok another update, I managed to resolve the issue. Apparently by changing Keyframes to 1-sec instead of Automatic it now works again.
My resolution is 3440x1440 and here's my other settings:

I don't really understand why but this seems to be the fix consistently.

View full answer

devedse · 2023-05-17T12:50:13Z

devedse
May 17, 2023
Author

I did some more digging to see if all ports are open. So with this command I ran a server on the docker host:

pi@dockerpi:~ $ echo -e "1935/tcp\n3333/tcp\n3334/tcp\n3478/tcp\n9001/tcp\n9999/tcp\n3478/udp\n10000/udp\n10001/udp\n10002/udp\n10003/udp\n10004/udp\n10005/udp\n4000/udp" | xargs -P 0 -n1 bash -c 'p=${0#*/}; echo "Listening on ${0%/*}/$p"; nc -l${p:0:1^tcp} ${0%/*}'
Listening on 1935/tcp
Listening on 3333/tcp
Listening on 3334/tcp
Listening on 3478/tcp
Listening on 9001/tcp
Listening on 9999/tcp
Listening on 3478/udp
Listening on 10000/udp
Listening on 10002/udp
Listening on 10001/udp
Listening on 10003/udp
Listening on 10004/udp
Listening on 10005/udp
Listening on 4000/udp

Then with this command I tried to connect to it and send some data:

echo -e "1935/tcp\n3333/tcp\n3334/tcp\n3478/tcp\n9001/tcp\n9999/tcp\n3478/udp\n10000/udp\n10001/udp\n10002/udp\n10003/udp\n10004/udp\n10005/udp\n4000/udp" | while read line; do p=${line#*/}; echo "Sending to ${line%/*}/$p"; echo "Reached port ${line%/*} on protocol $p" | nc -q 0 -${p:0:1} ***********.duckdns.org ${line%/*}; sleep 1; done

I can't really explain why but all ports work consistently except port 10001, this is the output of the server after running the client command:

Reached port 1935 on protocol tcp
Reached port 3333 on protocol tcp
Reached port 3334 on protocol tcp
Reached port 3478 on protocol tcp
Reached port 9001 on protocol tcp
Reached port 9999 on protocol tcp
Reached port 3478 on protocol udp
Reached port 10000 on protocol udp
Reached port 10002 on protocol udp
Reached port 10003 on protocol udp
Reached port 10004 on protocol udp
Reached port 10005 on protocol udp
Reached port 4000 on protocol udp

For now I've changed the OME config to use ports 10002/udp to 10005/udp since those are confirmed working.

Well either way, I've confirmed that all ports are open from the internet to the docker host in question.

0 replies

getroot · 2023-05-18T02:57:47Z

getroot
May 18, 2023
Maintainer Sponsor

When <TcpForce>true</TcpForce> it should play if the WebRTC Player can connect to your port 3334/tcp (Secured WebSocket for WebRTC Signaling) 3478/tcp (Internal TURN Server).

Is the connection to FriendIP:3334/tcp and FriendIP:3478/tcp good from outside and inside?

0 replies

devedse · 2023-05-18T07:44:16Z

devedse
May 18, 2023
Author

Hi @getroot ,

Yes it is. In my reply jusylt above this comment I've noted down all ports that are forwarded and how I verified if they actually are.

Anything else that could be going wrong? Can you maybe explain a bit around what part is not working for me? Because it seems at least some messages are flowing through.

Maybe it would help to compare the docker logs of a working server with mine?

Devedse

6 replies

devedse May 18, 2023
Author

That's definitely not the intention. I just tested it and you are right, the audio stream actually works 🤯 .
Do you know where the settings for that are?

For some reason I can't get it working in demo.ovenplayer.com, it just says Connection to low latency (OME) server failed. But as it is now working in my own player, what could be the fix for the audio things?

getroot May 18, 2023
Maintainer Sponsor

For me to know why your stream is working with audio only I have to check your Server.xml and ovenmediaengine.log files...

devedse May 18, 2023
Author

I've provided most of the server.xl in the initial post. Incase of the logs I've made screenshots. I'm not at home right now so I can't provide the full log but I'll give that later 😄

getroot May 18, 2023
Maintainer Sponsor

In Server.xml the part you didn't provide ( OutputProfile ) is the most important. Please refer to the manual.

https://airensoft.gitbook.io/ovenmediaengine/transcoding

devedse May 18, 2023
Author

Hmm I didn't change that part compared to previously when it worked. But basically I configured it to only pass through everything. I'll hope I can give you the logs later.

devedse · 2023-05-18T15:19:23Z

devedse
May 18, 2023
Author

Here's my full Server.xml:

<?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.ovenmediaengine.com:13478</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>2</WorkerCount>
                                </Signalling>

                                <IceCandidates>
                                        <IceCandidate>{FRIENDIP}:10002-10005/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>{FRIENDIP}: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>false</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>{FRIENDIP}:10002-10005/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>{FRIENDIP}: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>/etc/letsencrypt/live/webrtc.openstreamingplatform.com/cert.pem</CertPath>
                                <KeyPath>/etc/letsencrypt/live/webrtc.openstreamingplatform.com/privkey.pem</KeyPath>
                                <ChainCertPath>/etc/letsencrypt/live/webrtc.openstreamingplatform.com/fullchain.pem</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>***************.duckdns.org</Distribution>

                        <!-- Settings for multi ip/domain and TLS -->
                        <Host>
                                <Names>
                                        <Name>***************.duckdns.org</Name>
                                        <Name>{FRIENDIP}</Name>
                                        <Name>*</Name>
                                </Names>
                                <TLS>
                                        <CertPath>/etc/letsencrypt/live/***************.duckdns.org/cert.pem</CertPath>
                                        <KeyPath>/etc/letsencrypt/live/***************.duckdns.org/privkey.pem</KeyPath>
                                        <ChainCertPath>/etc/letsencrypt/live/***************.duckdns.org/fullchain.pem</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> -->

                        <!--
                        <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>ome-dev.airensoft.com</OriginHostName>
                        </OriginMapStore>
                        -->

                        <!-- 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>
                                                        <!--<Playlist>
                                                                <Name>for Webrtc</Name>
                                                                <FileName>abr</FileName>
                                                                <Options>
                                                                        <WebRtcAutoAbr>false</WebRtcAutoAbr>
                                                                </Options>
                                                                <Rendition>
                                                                        <Name>1080p</Name>
                                                                        <Video>1080p</Video>
                                                                        <Audio>opus</Audio>
                                                                </Rendition>
                                                                <Rendition>
                                                                        <Name>480p</Name>
                                                                        <Video>480p</Video>
                                                                        <Audio>opus</Audio>
                                                                </Rendition>
                                                                <Rendition>
                                                                        <Name>720p</Name>
                                                                        <Video>720p</Video>
                                                                        <Audio>opus</Audio>
                                                                </Rendition>
                                                                <Rendition>
                                                                        <Name>1080pVp8</Name>
                                                                        <Video>1080pVp8</Video>
                                                                        <Audio>opus</Audio>
                                                                </Rendition>
                                                                <Rendition>
                                                                        <Name>480pVp8</Name>
                                                                        <Video>480pVp8</Video>
                                                                        <Audio>opus</Audio>
                                                                </Rendition>
                                                                <Rendition>
                                                                        <Name>720pVp8</Name>
                                                                        <Video>720pVp8</Video>
                                                                        <Audio>opus</Audio>
                                                                </Rendition>

                                                        </Playlist> -->

                                                        <!--
                                                        You can provide ABR with Playlist. Currently, ABR is only supported in LLHLS.

                                                        <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>
                                                                <Audio>
                                                                        <Name>bypass_audio</Name>
                                                                        <Bypass>true</Bypass>
                                                                </Audio>
                                                                <Video>
                                                                        <Name>bypass_video</Name>
                                                                        <Bypass>true</Bypass>
                                                                </Video>
                                                                <Audio>
                                                                        <Codec>opus</Codec>
                                                                        <Bitrate>128000</Bitrate>
                                                                        <Samplerate>48000</Samplerate>
                                                                        <Channel>2</Channel>
                                                                </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 />
                                                <RTMP />
                                                <SRT />
                                                <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>4000</Port>
                                                                </Stream>
                                                        </StreamMap>
                                                </MPEGTS>
                                                <RTSPPull />
                                                <WebRTC>
                                                        <Timeout>30000</Timeout>
                                                </WebRTC>
                                        </Providers>
                                        <Publishers>
                                                <AppWorkerCount>1</AppWorkerCount>
                                                <StreamWorkerCount>2</StreamWorkerCount>
                                                <OVT />
                                                <WebRTC>
                                                        <Timeout>30000</Timeout>
                                                        <Rtx>false</Rtx>
                                                        <Ulpfec>false</Ulpfec>
                                                        <JitterBuffer>true</JitterBuffer>
                                                </WebRTC>
                                                <LLHLS>
                                                        <ChunkDuration>0.5</ChunkDuration>
                                                        <SegmentDuration>6</SegmentDuration>
                                                        <SegmentCount>10</SegmentCount>
                                                        <CrossDomains>
                                                                <Url>*</Url>
                                                        </CrossDomains>
                                                </LLHLS>
                                        </Publishers>
                                </Application>
                        </Applications>
                </VirtualHost>
        </VirtualHosts>
</Server>

2 replies

getroot May 19, 2023
Maintainer Sponsor

There is nothing wrong with your settings. It looks like your encoder is sending only audio. Please tell me your encoder information and how you set it up. I'll try to reproduce it.

devedse May 19, 2023
Author

Hmm, I'm using OSP (https://github.com/Open-Streaming-Platform/open-streaming-platform) which has a plugin to 'restream` streams to OME so you can have the low latency of WebRTC. Previously all of this worked fine but after the move to my new house it doesn't anymore.
It looks like that for some reason OSP isn't re-sending the audio then. (The HLS stream from OSP does contain the video stream).

devedse · 2023-05-19T08:09:08Z

devedse
May 19, 2023
Author

Ok another update, I managed to resolve the issue. Apparently by changing Keyframes to 1-sec instead of Automatic it now works again.
My resolution is 3440x1440 and here's my other settings:

I don't really understand why but this seems to be the fix consistently.

0 replies

devedse · 2023-05-19T17:39:01Z

devedse
May 19, 2023
Author

Is there a way to figure out if the client is using UDP or TCP btw?

0 replies

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OME worked before, but can't get it working behind double NAT with Tunnel and VPN to route traffic #1238

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