narayana/anotherim
narayana/anotherim
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Author SHA1 Message Date
Daniel Gultsch a9dd5a3c76 support sasl/temporary-auth-failure 
if the server is unable to query the database throwing a temporary-auth-failure
might be more appropriate
 2022-06-14 08:39:58 +02:00
Stephen Paul Weber 78048bbd3d Enable WebRTC-BindUsingInterfaceName/Enabled/ 
This makes 464XLAT networks (such as T-Mobile LTE) work.

https://bugs.chromium.org/p/webrtc/issues/detail?id=10707
 2022-03-10 16:29:00 +01:00
Daniel Gultsch 372078629b fix ice candidate sending when different credentials are used 2022-02-25 17:26:36 +01:00
Daniel Gultsch 1f772df74f remove security check that ensures rtp connection was properly finished 
this only causes race conditions
 2022-02-25 16:24:16 +01:00
Daniel Gultsch d6be6ddd18 use full file name for all new files 2022-02-22 16:05:02 +01:00
Daniel Gultsch d7f38a3e5a fix precondition for timeout handling 2022-02-12 10:19:54 +01:00
Daniel Gultsch b6442c0bd4 add Samsung S4 to hardware aec blacklist 
fixes #4267
 2022-01-18 11:30:23 +01:00
Daniel Gultsch db834a1f07 indicate call reconnect in notification 2021-11-19 12:26:11 +01:00
Daniel Gultsch f8a94161db don't play tone going from connect->reconnect->connect 2021-11-19 12:25:27 +01:00
Daniel Gultsch a508a81553 externalize rtc config generation into seperate method 2021-11-17 11:33:15 +01:00
Daniel Gultsch 61fb38cd84 clean up some error handling error ICE restarts 2021-11-17 10:49:16 +01:00
Daniel Gultsch 0a18c8613f assume credentials are the same for all contents when restarting ICE 2021-11-16 17:08:34 +01:00
Daniel Gultsch abb671616c synchronize setDescription calls 2021-11-16 15:17:12 +01:00
Daniel Gultsch 297a843b9c use implicit rollback (needed to be enabled on libwebrtc) 2021-11-16 13:17:10 +01:00
Daniel Gultsch 0698fa0d8c store peer dtls setup for later use in ice restart 2021-11-16 11:21:11 +01:00
Daniel Gultsch 70b5d8d81a set proper peer dtls setup on ice restart received 2021-11-15 21:49:31 +01:00
Daniel Gultsch 0a3947b8e3 terminate with application failure when failing to apply ICE restart 
This is fairly unlikely to happen in practice
 2021-11-15 17:18:45 +01:00
Daniel Gultsch 3f402b132b respond with tie-break to prevent ICE restart race 2021-11-15 13:03:04 +01:00
Daniel Gultsch 5b80c62a63 treat transport-info w/o candidates and changed credentials as offer 2021-11-14 18:22:18 +01:00
Daniel Gultsch 717c83753f delay discovered ice candidates until JingleRtpConnection is ready to receive 
otherwise setLocalDescription and the arrival of first candidates might race
the rtpContentDescription being set
 2021-11-11 21:02:17 +01:00
Daniel Gultsch b6dee6da6a reverse: webrtc: include oldState in onConnectionChange 
turns out we don’t need it and a better way is for RtpConnection to keep track of *all*
states in the current generation
 2021-11-11 17:05:36 +01:00
Daniel Gultsch 9c3f55bef2 use stopwatch to keep track of jingle rtp session duration 2021-11-11 16:52:18 +01:00
Daniel Gultsch 9843b72f6f always use Camera2Enumerator 2021-11-11 15:23:45 +01:00
Daniel Gultsch 61851e5f84 do not automacially hang up failed webrtc sessions 2021-11-11 14:40:15 +01:00
Daniel Gultsch 4ec0996dff webrtc: include oldState in onConnectionChange 2021-11-11 11:19:37 +01:00
Daniel Gultsch fda45a7c86 use implicit descriptions for WebRTC 
using the parameter-free form of setLocalDescription() solves some potential race conditions
that can occur - especially once we introduce restartIce()
 2021-11-10 16:40:24 +01:00
Daniel Gultsch bae9fc45c2 make rtcpMux optional 2021-10-31 10:20:58 +01:00
Daniel Gultsch 20e4d108d4 fixed regression of not handling jingle content map parsing failures 2021-10-05 15:43:05 +02:00
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 ac7855a332 show domains in manual cert accept dialog 2021-05-03 08:28:03 +02:00
Daniel Gultsch bc58fb0fbd Always verify hostname/domain 
There might be corner cases where it is required to use self signed
certificates. However there should be no corner cases where it is
required to use a wrong domain name. This commit swaps out the
MemorizingHostnameVerifier that let users accept wrong domains with the
standard XmppDomainVerifier.

closes #4066
 2021-04-30 09:55:22 +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 ce7f59a76c use okhttp to fetch captcha 2021-03-22 10:39:53 +01:00
Daniel Gultsch 1cd95aefa6 migrate redirection urls to HttpUrl 2021-03-22 10:12:53 +01:00
Daniel Gultsch 739d20428a optimize imports 2021-03-21 21:39:04 +01:00
Daniel Gultsch a6244d986a use settable futures for slot requester 2021-03-21 20:45:26 +01:00
Daniel Gultsch 8ac97b0027 disable extmap_allow_mixed by default 2021-03-21 19:40:52 +01:00
Daniel Gultsch e217551a82 migrate to OkHttp instead of HttpUrlConnection 
OkHttp gives us more fine grained control over the HTTP library and frees us from any platform bugs
 2021-03-19 14:57:20 +01:00
Daniel Gultsch b09a1432a3 Stanza.getErrorCondation only ever needs the tag name 2021-03-18 11:35:41 +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 b6d62c13ef use ascii notation for punycode domains in SNI 2021-02-07 09:38:55 +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 0569febf67 minor code clean up in XmppConnection class 2020-12-31 10:27:06 +01:00
Daniel Gultsch 0e54d8a2cf implement SCRAM-SHA512 2020-12-31 09:32:05 +01:00
Daniel Gultsch f23016c967 offline presences aborts session proposals. fixes #3943 2020-12-22 17:50:26 +01:00
Daniel Gultsch 1f392a688d initial (untested) support for easy onboarding invites 2020-12-01 20:31:30 +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
Daniel Gultsch bf85a55930 catch NPE when detecting camera facing. fixes #3820 2020-07-09 20:11:09 +02:00
Daniel Gultsch 6a6c9fb3bf ignore race condition when toggling fixes #3822 2020-07-09 19:14:28 +02:00
Daniel Gultsch fada3a63c9 store entire transport info for after session was accepted. fixes #3790 2020-06-22 18:07:27 +02:00
Daniel Gultsch dddb7ece25 show app failure instead of crashing when egl fails to init. fixes #3795 2020-06-18 20:37:56 +02:00
Daniel Gultsch 169ee99afa do not attempt to reject call if session had already ended. fixes #3798 2020-06-18 20:32:58 +02:00
Daniel Gultsch 7bcb29c482 be more liberal in 0167 payload-type parameter parsing 
some implementations will transform the following SDP coming from Firefox

m=audio 12346 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

to

<payload-type channels="1" name="telephone-event" clockrate="8000" id="101">
  <parameter value="0-15" xmlns="urn:xmpp:jingle:apps:rtp:1"/>
</payload-type>

While a missing name attribute is not legal according to the XEP; and 0-15 are
technically not just one value the following commit will accept it if there is
just one paramater.
 2020-06-17 21:15:12 +02:00
Daniel Gultsch c8f23aef4e error response to sending the jingle ft hash should not file the transfer 2020-06-15 21:33:32 +02:00
Daniel Gultsch ccdc91a497 remove check that would ensure you use jingle only with full jids 2020-06-14 09:01:47 +02:00
Daniel Gultsch 400c8461fc fix feature discovery in jingle file transfer for empty resources 2020-06-13 22:53:24 +02:00
Daniel Gultsch 0ba4892d3e RTP: write log message on background thread 2020-06-12 09:08:09 +02:00
Daniel Gultsch 644ad99520 create rtp end user state for connection lost. fixes #3769 2020-06-12 07:57:11 +02:00