sfu | future proof , experimental WebRTC VP9 SVC SFU wit end | Runtime Evironment library

The regex delimiters delimit the following parts:

The issue is that Pion (or any WebRTC implementation) is only aware of IP address it is listening on. It can't be aware of all the address that map/forward to it. People will also call this the Public IP or NAT Mapping. So when Pion emits it candidates they will probably look like 10.10.0.* and the remote peer will be unable to contact that.

What you should do is use the SettingEngine and set SetNat1To1IPs. If you know the public IP of the host it will rewrite the candidates with the public IP.

ICE is a tricky process. To understand it conceptually WebRTC for the Curious#Networking may be helpful. Will make sure to answer any follow up questions on SO quickly!

Keep the for loop, but use enumerate() there.

You're not appending to the list that's in the dictionary. [].append(line) creates a temporary list, appends to it, and then returns None, and this None value is put into the dictionary.

You can use a collections.defaultdict for this, to create a dictionary element that contains a list as needed.

Why there is no second call for the CAS server for ticket validation?

There is. The second call is a back-channel call from your application server over to the CAS server. By definition, this is not something you would see in your browser. This call goes over to the CAS server behind the scenes to validate the service ticket received in the first leg (i.e. ST-xyz). The Java CAS client library should be automatically doing this for you, and you can verify this in the logs.

If you don't see this happening, your configuration is not set correctly or there is an error along the process.

Is it hidden?

Hidden from the browser, as it's a back-channel call. For additional details on what happens and why, please study the CAS protocol.

How do I extract user and authtype attributes from the XML document and store them in the database?

The Java CAS client library typically extracts the user id and other attributes. Then, the user-id would be available under the REMOTE-USER header that can be fetched via the http request object. If you have access to the http session, you can also fetch the final Assertion from the session which contains the CAS payload:

Yes the connection between a client and an SFU is a WebRTC connection. The SFU is acting as a peer end-point.

There's no distinction made in the w3c specification between a peer and an SFU, except it leaves out defining reception of simulcast in browser-based clients, something you'll need an SFU (or MCU) for.

1. Do a regular offer/answer exchange between each client and the SFU end-point over signaling.
   • If the server is the offerer it sends an offer to receive simulcast to the client. A normal offer, except the client can tell after setRemoteDescription it will be sending multiple encodings (layers) of a single video track:

run the command:

sudo nano /etc/bigbluebutton/nginx/sip.nginx

something like this will open:

After discussing the matter with Microsoft, it was confirmed that this specific case is not supported by Microsoft AD by design. Specifically if front end service is in one forest, and the user and resource service are in a different forest, then this use case is not supported.

Hopefully this helps for someone that encounter this issue in the future.

Not without alteration.

Internally WebRTC's internal audio/video pipelines are directly tied to encoder/decoders.

PeerConnectionFactory allows you to provide a video decoder/encoder factory, so you can short circuit the logic here, and grab the encoded frames, mock up a stream, and feed them directly into it as a relay, creating a new PeerConnection and setting those streams onto it.

The audio end is more difficult. There isn't a codec factory, so you will have to short circuit the logic there probably by alteration of libwebrtc.

The final question is RTCP termination, and how to override the mechanisms for quality/bandwidth control to not create a "One goes out, they all go out." situation.

Since libwebrtc will be the SFU, it will receive RTCP feedback from its remote peer for the content it is proxying, and vice versa.

For a 1-1 situation, it needs to be able to forward the RTCP feedback to the remote peer.

For multipoint, it needs to perform some logic to determine if one of the peers is problematic, and stop sending it video, switch off its video feed, or attempt to switch to a lower bitrate video stream. Basically it needs to act as a conduit that attempts to predict why/how packet loss is occurring, and keep as many audio/video feeds operating normally at at the highest possible quality for each peer.

How exactly to hijack the RTCP feedback mechanisms in libwebrtc, I think that again will likely require some customization/hooks into libwebrtc