Commit graph

348 commits

Author SHA1 Message Date
Daniel Gultsch 3ede2d00bd remove logging 2021-10-02 16:54:19 +02:00
Daniel Gultsch d2a387e82f correctly calculate socks destination 2021-10-02 16:44:36 +02:00
Daniel Gultsch da14f83a42 ensure all bytes are read in socks handshake. fixes #4188 2021-10-02 14:24:36 +02:00
Daniel Gultsch 63f5f8c89d modify TODOs in JingleRtpConnection upon better understanding of the WebRTC stack 2021-09-08 10:47:34 +02:00
Daniel Gultsch 7466d12505 ring during device discovery 2021-05-22 19:37:20 +02:00
Daniel Gultsch 87f99d3570 Transferables interface needs to differentiate between 0 and null file size 2021-05-17 15:51:21 +02:00
Daniel Gultsch 67e5f839f1 ignore crypto callbacks when rtp session has already been terminated 2021-05-08 11:50:18 +02:00
Daniel Gultsch 9182a300c5 report fingerprint missmatch as securiy exception 2021-05-08 10:35:07 +02:00
Daniel Gultsch 8d391753d7 encrypt rtp map as future 2021-05-08 08:45:31 +02:00
Daniel Gultsch 337aa4a110 consider Config.REQUIRE_RTP_VERIFICATION on decrypt. fail as future 2021-05-07 22:55:20 +02:00
Daniel Gultsch ddf597e0d3 invoke x509 verification upon receiving prekey message in rtp session 2021-05-06 18:40:35 +02:00
Daniel Gultsch e2324209ed make sure omemo sessions are verified if the the respective config flag is set 2021-05-04 19:04:01 +02:00
Daniel Gultsch 48156dd27f a/v calls: seperate out SECURITY error from APP_FAILURE
until now problems with verifying the call (omemo or DTLS missing) would
just be another app failure. This commit displays verifications problems as
their own thing.
2021-05-04 10:10:34 +02:00
Daniel Gultsch 6d91551f59 use onAddTrack instead of deprecated onAddStream 2021-05-03 13:06:42 +02:00
Daniel Gultsch 0717f9ba18 upgrade libwebrtc to m90 and enable extmap-allow-mixed 2021-05-03 09:48:46 +02:00
Daniel Gultsch 9fc04c4b1e when receiving out-of-order session-init in terminal state do not move to terminal again
fixes #4049
2021-04-08 10:23:39 +02:00
Daniel Gultsch 30c9e7399e log track class in onAddTrack 2021-03-29 10:57:56 +02:00
Daniel Gultsch 1822a71c2a Do not crash when receiving video call on device w/o camera
Upon accepting a video call on a device that can not establish a video track on
its own (for example by not having a camera), displaying the video enable/disable
button would fail. This commit defaults this button to disabled.
2021-03-26 12:54:26 +01:00
Daniel Gultsch 739d20428a optimize imports 2021-03-21 21:39:04 +01:00
Daniel Gultsch 8ac97b0027 disable extmap_allow_mixed by default 2021-03-21 19:40:52 +01:00
Daniel Gultsch 6f1b71970d parse extmap-allow-mixed 2021-03-16 18:52:38 +01:00
Daniel Gultsch 3baacf8862 switch to unified plan 2021-03-16 18:52:38 +01:00
Daniel Gultsch 2681ad82e1 complain if mLineIndex can not be found when receiving candidates 2021-03-16 18:52:25 +01:00
Christopher Vollick ef24d2050b Remove Renomination from WebRTC Options
This is a feature of WebRTC that's [not standardized][1] and only
supported by libwebrtc. Since there's no support in jingle for passing
this capability from one peer to another, we're currently hard-coding
this option into both the local candidate and also the remote candidate
so they can use it.

But I'm trying to call a user that isn't using WebRTC, and renomination
is causing the call to stay in "connecting..." state for 10 or 20
seconds, sometimes longer, while both sides wait for the other to
nominate something based on their individual beliefs about the standards
they're using.

Removing this seems to make connecting relatively instantaneous.

If we want to reintroduce this feature, we should probably make a XEP so
the peers can negotiate honestly about it, and only use it if both sides
truely support the feature.

[1]: https://datatracker.ietf.org/doc/html/draft-thatcher-ice-renomination-01
2021-03-04 08:26:52 +00:00
Daniel Gultsch c5f801c1fe do not push empty candidates to backlog 2021-03-03 13:12:10 +01:00
Daniel Gultsch d52c46d582 use omemo verification only if omemo is enabled in conversation 2021-03-03 12:55:27 +01:00
Daniel Gultsch 3ee70b1d48 show verified shield in rtp session activity 2021-03-03 09:41:05 +01:00
Daniel Gultsch e4b2bb4a42 throw exception when unable to encrypt 2021-03-03 08:22:21 +01:00
Daniel Gultsch 8a6430ae29 ground work for omemo dtls verification 2021-03-02 21:13:49 +01:00
Daniel Gultsch f92ea5c70b resend <propose/> only if server has stream mgmt 2021-02-21 13:37:08 +01:00
Daniel Gultsch 484f633180 let Conversations (not Android) play ringtone and vibration
fixes #3972 fixes #3801 fixes #3931
2021-02-18 20:55:31 +01:00
Daniel Gultsch 72e268e6b1 add TODO comments wrt to missing <retract/> parsing 2021-02-18 09:36:51 +01:00
Daniel Gultsch db447f845e resend session proposal on rebind 2021-02-12 11:36:44 +01:00
Daniel Gultsch 6cab0ad496 make rtp proposal tracked by SM. fixes #3983 2021-02-12 10:35:13 +01:00
Daniel Gultsch 7330d8a7f0 fixed race conditions around PROCEED state. fixes #3989 2021-02-11 16:56:57 +01:00
Daniel Gultsch 0a2c753620 do not use offline fallback rtp capability if account is disabled 2021-01-26 09:35:03 +01:00
Daniel Gultsch 8ce7bfb95e automated code clean up 2021-01-23 09:25:34 +01:00
Daniel Gultsch e711b3d294 remember last rtp capability 2021-01-22 08:24:19 +01:00
Ferdinand Pöll 453ca7c0ed Migrate from Android Support Library to AndroidX
Unignored gradle.properties since androidX requires additions there
See also https://developer.android.com/jetpack/androidx/migrate
2021-01-18 20:49:35 +01:00
Daniel Gultsch 372ddbfb49 Revert "offline presences aborts session proposals. fixes #3943"
This reverts commit f23016c967.
2021-01-06 09:03:42 +01:00
Emmanuel Gil Peyrot 17c697eed9 add 'id' attribute to outgoing ICE-UDP candidates
this attribute is mandatory as per the XEP.
2021-01-03 16:32:28 +00:00
Daniel Gultsch 2bec5459c5 properly null check ufrag and pwd before whitespace checking. fixes #3956 2021-01-03 16:05:17 +01:00
Daniel Gultsch f23016c967 offline presences aborts session proposals. fixes #3943 2020-12-22 17:50:26 +01:00
Daniel Gultsch d158eeaf72 terminate jingle call when regular call starts 2020-08-24 12:47:54 +02:00
Daniel Gultsch 91e94db747 extend isBusyState to check phone state as well 2020-08-24 09:51:26 +02:00
Daniel Gultsch 15b323ee69 fix crash after session-accept failed and session-accept contained candidates
Conversations would attempt to feed any candidates found in the session-accept into
WebRTC; even if the session wasn’t setup correctly.

this commit processes the candidates only if the session was setup correctly

fixes #3867
2020-08-22 08:12:28 +02:00
Daniel Gultsch 637c0cb15a fixed rare race condition when receiving transport info right after WebRTCWrapper closes
fixes #3849
2020-08-01 14:18:03 +02:00
Daniel Gultsch 1ae7d6be16 recover from pre-jingle connection states (discover etc) into full fledged jingle connection
fixes #3847
2020-08-01 09:50:54 +02:00
Daniel Gultsch f22e33e3ea fixed race condition of WebRTCWrapper being closed before transitioning into terminal state
JingleRTPConnection shuts down the WebRTCWrapper before transitioning into a terminal state.
(This allows us to make sure it is actually closed when reaching that state);
However that means that, when we get a UI redrawn inbetween closing and transitioning we might
still be in SESSION_ACCEPTED but with no PeerConnection. This traditionally has triggered
an IllegalStateException on getting the EndUserState.
This commit catches the ISE and returns 'ENDING' instead.
Chances are that this is only visibiliy for a very brief time in the UI before the transition
triggers the UI to redraw with the proper state.

fixes #3848
2020-08-01 08:20:10 +02:00
Daniel Gultsch 32d55346cc ensure server triggered jingle iq-errors get routed properly 2020-07-18 16:14:39 +02:00