[Webkit-unassigned] [Bug 233414] New: A server-reflexive candidate is discarded when the host IPv4 address is public and obfuscated

Sun Nov 21 22:28:54 PST 2021

https://bugs.webkit.org/show_bug.cgi?id=233414

            Bug ID: 233414
           Summary: A server-reflexive candidate is discarded when the
                    host IPv4 address is public and obfuscated
           Product: WebKit
           Version: Safari 15
          Hardware: Mac (Intel)
                OS: macOS 12
            Status: NEW
          Severity: Normal
          Priority: P2
         Component: WebRTC
          Assignee: webkit-unassigned at lists.webkit.org
          Reporter: gorisanson at gmail.com
                CC: youennf at gmail.com

Created attachment 444945

  --> https://bugs.webkit.org/attachment.cgi?id=444945&action=review

Result screenshot of taking the steps to reproduce the bug and corresponding to the Web Inspector Console Logs

A server-reflexive (srflx) candidate is discarded when the host IPv4 address is public and obfuscated.

According to https://www.rfc-editor.org/rfc/rfc8445#section-5.1.3, the srflx candidate is considered redundant and eliminated when the host address is public (and not obfuscated). In this case, WebRTC P2P connection can be achieved since the host address is public and the same as the srflx candidate address.

However, when the host address is both public and obfuscated, according to https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-mdns-ice-candidates-02#section-3.1.2.2, a srflx candidate MUST NOT be considered redundant and so must not be eliminated. But, it seems that Safari eliminates the srflx candidate when the host address is both public and obfuscated.

Here are steps to reproduce the bug:
1. Connect to the Internet directly with a public IP (behind no NAT)
2. Open the Trickle ICE sample in Safari: https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/
3. Click "Gather candidates"

The following is logs of Web Inspector Console when I take the steps above, with setting WebRTC Logging "Verbose" (Web Inspector -> Settings -> Console -> WebRTC Logging).

Safari Web Inspector Console Logs with WebRTC Logging "Verbose":
--------
[Log] Creating new PeerConnection with config={"iceServers":[{"urls":["stun:stun.l.google.com:19302"]}],"iceTransportPolicy":"all","iceCandidatePoolSize":"0"} (main.js, line 157)
[Log] RTCPeerConnection::RTCPeerConnection(6FABB4D9) 
[Info] RTCPeerConnection::initializeConfiguration(6FABB4D9) 
[Info] RTCPeerConnection::addTransceiver(6FABB4D9) 
[Log] MediaStreamTrackPrivate::MediaStreamTrackPrivate(C4B26C3DDFE44905) 
[Log] RealtimeIncomingAudioSource::setLogger(C4B26C3DDFE44905) Audio, remote audio, 
[Log] MediaStreamTrack::MediaStreamTrack(C4B26C3DDFE44905) 
[Log] RTCPeerConnection::createOffer(6FABB4D9) 
[Log] PeerConnectionBackend::createOfferSucceeded(6FABB4D9) Create offer succeeded:
v=0
o=- 3826257032639583107 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:rO9Z
a=ice-pwd:3rHfBci/3naaqdEDIOQPNYD/
a=ice-options:trickle
a=fingerprint:sha-256 13:38:36:50:7C:9D:01:68:CB:FC:91:BE:BD:28:FB:AC:49:67:A9:5E:CD:09:5F:BC:61:D8:8E:AF:2B:85:AA:F1
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:- f38bcd1c-1d2c-4fe1-8557-3aabe5a2f352
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3914177754 cname:Ed56B/h1MvTWyenf
a=ssrc:3914177754 msid:- f38bcd1c-1d2c-4fe1-8557-3aabe5a2f352
a=ssrc:3914177754 mslabel:-
a=ssrc:3914177754 label:f38bcd1c-1d2c-4fe1-8557-3aabe5a2f352

[Log] RTCPeerConnection::setLocalDescription(6FABB4D9) Setting local description to:
v=0
o=- 3826257032639583107 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:rO9Z
a=ice-pwd:3rHfBci/3naaqdEDIOQPNYD/
a=ice-options:trickle
a=fingerprint:sha-256 13:38:36:50:7C:9D:01:68:CB:FC:91:BE:BD:28:FB:AC:49:67:A9:5E:CD:09:5F:BC:61:D8:8E:AF:2B:85:AA:F1
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:- f38bcd1c-1d2c-4fe1-8557-3aabe5a2f352
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3914177754 cname:Ed56B/h1MvTWyenf
a=ssrc:3914177754 msid:- f38bcd1c-1d2c-4fe1-8557-3aabe5a2f352
a=ssrc:3914177754 mslabel:-
a=ssrc:3914177754 label:f38bcd1c-1d2c-4fe1-8557-3aabe5a2f352

[Log] RTCPeerConnection::setSignalingState(6FABB4D9) have-local-offer
[Info] RTCPeerConnection::dispatchEvent(6FABB4D9) dispatching 'signalingstatechange'
[Log] PeerConnectionBackend::setLocalDescriptionSucceeded(6FABB4D9) 
[Log] RTCPeerConnection::updateIceGatheringState(6FABB4D9) gathering
[Info] RTCPeerConnection::dispatchEvent(6FABB4D9) dispatching 'icegatheringstatechange'
[Info] RTCPeerConnection::updateConnectionState(6FABB4D9) state changed from: new to connecting
[Info] RTCPeerConnection::dispatchEvent(6FABB4D9) dispatching 'connectionstatechange'
[Info] RTCPeerConnection::dispatchEvent(6FABB4D9) dispatching 'negotiationneeded'
[Log] PeerConnectionBackend::newICECandidate(6FABB4D9) Gathered ice candidate:candidate:484232637 1 udp 2122260223 175.212.38.195 60853 typ host generation 0 ufrag rO9Z network-id 1 network-cost 50
[Info] RTCPeerConnection::dispatchEvent(6FABB4D9) dispatching 'icecandidate'
[Log] PeerConnectionBackend::newICECandidate(6FABB4D9) Gathered ice candidate:candidate:1381970253 1 tcp 1518280447 175.212.38.195 9 typ host tcptype active generation 0 ufrag rO9Z network-id 1 network-cost 50
[Info] RTCPeerConnection::dispatchEvent(6FABB4D9) dispatching 'icecandidate'
[Log] PeerConnectionBackend::doneGatheringCandidates(6FABB4D9) Finished ice candidate gathering
[Info] RTCPeerConnection::dispatchEvent(6FABB4D9) dispatching 'icecandidate'
[Log] RTCPeerConnection::updateIceGatheringState(6FABB4D9) complete
[Info] RTCPeerConnection::dispatchEvent(6FABB4D9) dispatching 'icegatheringstatechange'
[Log] MediaStreamTrack::stopTrack(C4B26C3DDFE44905) 1
[Log] MediaStreamTrackPrivate::endTrack(C4B26C3DDFE44905) 
[Log] MediaStreamTrackPrivate::updateReadyState(C4B26C3DDFE44905) Ended
[Log] MediaStreamTrackPrivate::preventSourceFromStopping(C4B26C3DDFE44905) true
[Log] RealtimeIncomingAudioSource::requestToEnd(C4B26C3DDFE44905) 
[Log] RealtimeIncomingAudioSource::end(C4B26C3DDFE44905) 
[Log] MediaStreamTrack::trackEnded(C4B26C3DDFE44905) 
[Log] MediaStreamTrack::trackEnded(C4B26C3DDFE44905) firing 'ended' event
[Log] RTCPeerConnection::setSignalingState(6FABB4D9) stable
[Info] RTCPeerConnection::dispatchEvent(6FABB4D9) dispatching 'signalingstatechange'
---------

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