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Integrate CSP access control into algorithms #36

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alvestrand opened this issue Jan 19, 2018 · 9 comments
Closed

Integrate CSP access control into algorithms #36

alvestrand opened this issue Jan 19, 2018 · 9 comments
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@alvestrand
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If w3c/webappsec-csp#287 lands in the CSP spec, the webrtc spec should specify where and how this access is checked.

As per comment in that thread, suggestion:

The following situations are to be checked according to this directive:

  • A host URL occurs in the list of RTCIceServers of an RTCConfiguration when a PeerConnection is created. In this case, the PeerConnection creation will fail.
  • A host URL occurs in the list of RTCIceServers of an RTCConfiguration when a PeerConnection's setConfiguration method is called. In this case, setting the configuration will fail.
  • An address occurs in the ip, protocol and port fields of an RTCIceCandidate created from SetRemoteDescription or AddIceCandidate. In this case, the call will be rejected.

And perhaps a note of caution, something like: "Due to the problem of listing all possible communication partners for a WebRTC application, the "*" value is likely to be the most useful value to set as the value of the "webrtc-src" directive".

@murillo128
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An address occurs in the ip, protocol and port fields of an RTCIceCandidate created from SetRemoteDescription or AddIceCandidate. In this case, the call will be rejected.

I will prefer to ignore that ICE candidate instead of dropping the whole call. That would allow to specify the TURN server ip on the CSP directive and get a P2P call going through it without needed to modify any existing code. If we drop the call, the SDP/candidates would need to be filtered out by the app to be able to do so.

@alvestrand alvestrand self-assigned this Jan 25, 2018
@alvestrand
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Referencing #1727 (cause bug) and w3c/webappsec-csp#287 (suggested CSP change) for context.

@alvestrand
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Discussion is ongoing at the webappsec-csp bug.

@dontcallmedom
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@alvestrand the discussion seems to have stalled in CSP; if we want this to be part of 1.0, I think it would be useful to rekindle progress on this

@alvestrand
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Since this is a new feature, and we've stopped adding new features, I'm moving this to the NV repo.

@alvestrand alvestrand transferred this issue from w3c/webrtc-pc Jul 4, 2019
@aboba
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aboba commented Jul 10, 2019

@alvestrand Does CSP require its own use case, or can we add a CSP requirement to another use case such as the Secure Conferencing Use Case PR #34 ?

@lgrahl
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lgrahl commented Jul 10, 2019

I'd classify this as a general issue. It can't really be tied to a specific use case.

@aboba
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aboba commented Jul 15, 2019

@lgrahl. Makes sense to not tie it to a specific use case. Will add it to the requirements list and cite it in multiple use cases.

aboba added a commit that referenced this issue Jul 15, 2019
Fix for Issues #35 and #36.
@aboba
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aboba commented Aug 27, 2019

With merger of PR 38, closing this issue.

@aboba aboba closed this as completed Aug 27, 2019
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