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Hello, I'm trying to write a program which makes use of the self defined webrtc ingest protocol to provide a webrtc stream to OME. I've had some success in receiving the offer and sending a response but I see references to SDP parsing errors and unsupported protocols in the server logs, however I can also see it appears to set up a stream. The SDP I receive from the server is as follows. I don't have much experience with WebRTC and its associated technologies but from what I understand I'll post my C# code at the end which handles the signalling, its not complex so should be easy to follow even with little C# experience. One thing I'm unusre of is if I should be adding tracks before setting the remote description, for now I am with hard coded codecs and format IDs based on what the SDP usually has for VP8. If I don't add these tracks I get additional errors in the server log, this is probably down to my lack of knowledge on WebRTC. C# // Setup buffer to read socket data into and a web socket to negotiate signalling.
// Custom verification callback to work around self signed cert.
using var buffer = MemoryPool<byte>.Shared.Rent(ushort.MaxValue);
using var socket = new ClientWebSocket();
socket.Options.RemoteCertificateValidationCallback = (object sender, X509Certificate? certificate, X509Chain? chain, SslPolicyErrors sslPolicyErrors) =>
{
return true;
};
// Connect to OME and request an SDP offer.
await socket.ConnectAsync(new Uri("wss://localhost:3334/app/bypass_stream?direction=send"), default);
await socket.SendAsync(Encoding.UTF8.GetBytes(JSONWriter.ToJson(new { command = "request_offer" })), WebSocketMessageType.Text, false, default);
var result = await socket.ReceiveAsync(buffer.Memory, default);
var sdpstr = Encoding.UTF8.GetString(buffer.Memory.Slice(0, result.Count).Span);
var offer = JSONParser.FromJson<Offer>(sdpstr);
// Create out peer with the provided ice servers and setup some listeners to track interesting events.
var config = new RTCConfiguration { iceServers = offer.ice_servers.Select(ice => new RTCIceServer { credential = ice.credential, username = ice.user_name, urls = string.Join(", ", ice.urls) }).ToList() };
var peer = new RTCPeerConnection(config);
// Add video and audio track.
// Should we be doing this now? I've just hard coded these based on the sort of SDPs we get.
peer.addTrack(new MediaStreamTrack(new VideoFormat(VideoCodecsEnum.VP8, 101), MediaStreamStatusEnum.SendOnly));
peer.addTrack(new MediaStreamTrack(new AudioFormat(AudioCodecsEnum.OPUS, 102, 48000, 2), MediaStreamStatusEnum.SendOnly));
foreach (var str in offer.candidates)
{
peer.addIceCandidate(new RTCIceCandidateInit { candidate = str.candidate, sdpMLineIndex = (ushort)str.sdpMLineIndex });
}
peer.setRemoteDescription(offer.sdp);
peer.VideoLocalTrack.RestrictCapabilities(new VideoFormat(VideoCodecsEnum.VP8, 101));
var answer = peer.createAnswer();
var response = new Offer { command = "answer", id = offer.id, sdp = answer };
var json = JSONWriter.ToJson(response);
await peer.setLocalDescription(answer);
await socket.SendAsync(Encoding.UTF8.GetBytes(json), WebSocketMessageType.Text, false, default);Hopefully that all makes enough sense to get some pointers on whats going wrong, any help greatly appreciated. Thanks. |
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Answered by
getroot
Feb 29, 2024
Replies: 1 comment 11 replies
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I wish I knew more about SDP (but it feels like I learn a bit every time this comes up). But I have one observation about the particular line that is giving the error: In OME the offer is: But your answer is: I think the F is important for feedback/error correction. A couple of possible red herrings but also things I noticed:
Good luck and keep us posted! |
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11 replies
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SAVP denotes SRTP and SAVPF denotes SRTCP. OME also works if it is a SAVP, so your SDP should be accepted. I patched it and committed it to the latest master. Could you check if your client is working? |
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Thanks, thats great! I'll give it a try tomorrow and see how it goes. |
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Latest master does fix the SDP errors I was seeing, thanks! Unfortunately my client isn't fully working, I'm hitting a new error. I'm now seeing an ICE warning about "agent has expired" and an error saying "102 channel to be delete cannot be found". as an extra point of comparison I tried to use OBS and the WHIP input, this also doesn't work, but I don't see any errors or warnings in the log, OBS just disconnects (using "http://localhost:3333/app/bypass_stream?direction=whip" as the url). So maybe there's something more fundamentally wrong with my server config. |
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This is a situation where signaling is complete, but ICE connection is not established. *:10000-10004/udp If you are using * in the configuration, it would be a good idea to put your actual IP instead of *. I think your network configuration is also strange. Please upload Server.xml. You are probably using a virtual machine. OBS advertises their ICE Candidate as follows: And OME advertises their ICE Candidate as follows: I don't see any IP bands where OME and OBS can communicate with each other. |
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Answer selected by
Aidan63
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Try using an external server with a public IP to narrow down the possible factors |
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Thanks for the help. You were right about the wildcard in the configuration, I changed it to "127.0.0.1" in the WebRTC ice candidate provider and OBS and my client now successfully connect and can send frames! (ome is in a docker container and my client on my desktop PC) However... the player now seems to not connect to the stream. I'm seeing not much info in the server logs and in the web browser I see that it requests and offer and receives a response, but thats it.
(its connecting to "ws://127.0.0.1:3333/app/stream") I've had the player working before where I'm producing an rtmp stream and pulling a webrtc stream, but not producing a webrtc stream. That was with the official ome docker image and no changes to the config file, not the master based one I've got now. So I'll have to do a comparison to see whats different between my one and the one provided. (github doesn't let you attach xml files for some reason...) <?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 IPv4 IPs, "::" means all IPv6 IPs) -->
<!-- Uncomment the line below to enable IPv6 -->
<!-- <IP>::</IP> -->
<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>
<!--
Below is an example of listening 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>4000-4004,4005/udp</Port>
-->
<Port>4000/udp</Port>
</MPEGTS>
<WebRTC>
<Signalling>
<Port>3333</Port>
<TLSPort>3334</TLSPort>
<WorkerCount>1</WorkerCount>
</Signalling>
<IceCandidates>
<!-- Uncomment the line below to use IPv6 ICE Candidate -->
<!-- <IceCandidate>[::]:10000-10004/udp</IceCandidate> -->
<IceCandidate>127.0.0.1:10000-10004/udp</IceCandidate>
<!-- Uncomment the line below to use a link local address when specifying an address with IPv6 wildcard (::) -->
<!-- <EnableLinkLocalAddress>true</EnableLinkLocalAddress> -->
<!--
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.
-->
<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>
<!-- Uncomment the line below to use IPv6 ICE Candidate -->
<!-- <IceCandidate>[::]:10000-10004/udp</IceCandidate> -->
<IceCandidate>*:10000-10004/udp</IceCandidate>
<!-- Uncomment the line below to use a link local address when specifying an address with IPv6 wildcard (::) -->
<!-- <EnableLinkLocalAddress>true</EnableLinkLocalAddress> -->
<!--
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.
-->
<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>
-->
<!--
Refer https://airensoft.gitbook.io/ovenmediaengine/alert
<Alert>
<Url></Url>
<SecretKey></SecretKey>
<Timeout>3000</Timeout>
<Rules>
<Ingress>
<MinBitrate></MinBitrate>
<MaxBitrate></MaxBitrate>
<MinFramerate></MinFramerate>
<MaxFramerate></MaxFramerate>
<MinWidth></MinWidth>
<MinHeight></MinHeight>
<MaxWidth></MaxWidth>
<MaxHeight></MaxHeight>
<MinSamplerate></MinSamplerate>
<MaxSamplerate></MaxSamplerate>
<LongKeyFrameInterval />
<HasBFrames />
</Ingress>
</Rules>
</Alert>
-->
<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> -->
<!--
<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>
-->
<!-- Default CORS Settings -->
<CrossDomains>
<Url>*</Url>
</CrossDomains>
<!-- Settings for applications -->
<Applications>
<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</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.
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.
-->
<!--
<Playlist>
<Name>WebRTC</Name>
<FileName>abr</FileName>
<Options>
<WebRtcAutoAbr>true</WebRtcAutoAbr>
</Options>
<Rendition>
<Name>High</Name>
<Video>high</Video>
</Rendition>
<Rendition>
<Name>Medium</Name>
<Video>medium</Video>
</Rendition>
<Rendition>
<Name>Low</Name>
<Video>low</Video>
</Rendition>
</Playlist>
<Encodes>
<Video>
<Name>high</Name>
<Codec>h264</Codec>
<Bitrate>30000000</Bitrate>
<Width>1920</Width>
<Height>384</Height>
<BFrames>0</BFrames>
<Preset>faster</Preset>
</Video>
<Video>
<Name>medium</Name>
<Bypass>true</Bypass>
<Bitrate>15000000</Bitrate>
<Width>1280</Width>
<BFrames>0</BFrames>
<Preset>faster</Preset>
</Video>
<Video>
<Name>low</Name>
<Bypass>true</Bypass>
<Bitrate>7500000</Bitrate>
<Width>800</Width>
<BFrames>0</BFrames>
<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>
<CrossDomains>
<Url>*</Url>
</CrossDomains>
</WebRTC>
<Schedule>
<MediaRootDir>/opt/ovenmediaengine/media</MediaRootDir>
<ScheduleFilesDir>/opt/ovenmediaengine/media</ScheduleFilesDir>
</Schedule>
</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>0.5</ChunkDuration>
<!-- <PartHoldBack> SHOULD be at least three times the <ChunkDuration> -->
<PartHoldBack>1.5</PartHoldBack>
<SegmentDuration>6</SegmentDuration>
<SegmentCount>10</SegmentCount>
<DVR>
<Enable>false</Enable>
<TempStoragePath>/tmp/ome_dvr/</TempStoragePath>
<MaxDuration>3600</MaxDuration>
</DVR>
<DRM>
<Enable>false</Enable>
<InfoFile>path/to/file.xml</InfoFile>
</DRM>
<CrossDomains>
<Url>*</Url>
</CrossDomains>
</LLHLS>
</Publishers>
</Application>
</Applications>
</VirtualHost>
</VirtualHosts>
</Server> |
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I think you missed that |
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Yes, I forgot about the publishers section. I've also updated them all to use my machines network IP instead of localhost, but still no luck. Thanks for the hint with the webrtc-internals chrome page, with that I can see there's a single "addIceCandidateFailed" |
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This is a situation where signaling is complete, but ICE connection is not established. *:10000-10004/udp If you are using * in the configuration, it would be a good idea to put your actual IP instead of *.
I think your network configuration is also strange. Please upload Server.xml. You are probably using a virtual machine.
OBS advertises their ICE Candidate as follows: