* Changes initialization of videoSIPGW.
* Adds some errors returned on creating videoSIPGW session.
* Fixes sending videoSIPGW session STATE_CHANGED event.
* Fixes sending jibriIQ, no result status is received.
* Adds VIDEO_SIP_GW_SESSION_STATE_CHANGED to JitsiConferenceEvents.
* Fixing comments.
* ref: Simplifies the logic for handling an incoming jingle session-initiate.
* fix: Don't redundantly log cross region
information under a field name called "label".
* cleanup: Simplifies code. Adds the userAgent as a permanent property
for statistics (so that the client doesn't have to).
* ref: Names the parameter which specifies the name of the event "eventName".
* ref: Extracts event names to AnalyticsEvents.
* ref: Exports and imports constants individually.
* fix: Fixes CONNECTION_TIMES event names.
* ref: Arranges constants alphabetically.
* ref: Adds line breaks.
* revert: Reverts be665cbff7.
* ref: Renames "peerjid".
* ref: Refactors the initialization of a peer connection.
* feat: Re-implements the A/B test for the "suspend video" feature.
* squash: Deep copy.
* ref: Renames forceSuspendVideo to abtestSuspendVideo.
Uses optional statsId to report to callstats and push it to presence. (#608)
* Uses optional statsId to report to callstats and push it to presence.
The feature is behind a flag which is disabled by default.
* Renames statsId to statsID.
* Fixes doc.
ESLint 4.8.0 discovers a lot of error related to formatting. While I
tried to fix as many of them as possible, a portion of them actually go
against our coding style. In such a case, I've disabled the indent rule
which effectively leaves it as it was before ESLint 4.8.0.
With JitsiAuthConnection the API consumer has to:
1. JitsiConference.createAuthenticationConnection,
2. JitsiAuthConnecition.authenticateAndUpgradeRole,
3. Wait on the Promise of 2,
4. Maybe cancel the JitsiAuthConnection of 1.
With authenticateAndUpgradeRole the API consumer has to:
1. JitsiConference.authenticateAndUpgradeRole,
2. Wait on the thenableWithCancel of 1.
3. Maybe cancel the thenableWithCancel of 1.
There are scenarios when it's ok to call setP2PStatus again with
the same value. For example when P2P is stopped, before it starts with
the A/B testing mode enabled. It's only important to know that it
happened, but it's not an error, because the code will not execute
and return immediately.
The 'ondatachannel' even listener must be set as soon as
the TraceablePeerConnection is created and before the offer is accepted.
We've observed situations where the ondatachannel event was missed,
because the listener is bound too late from the JingleSessionPC
'acceptOffer' success callback.
* ref: Moves the deployment info variables to config.js
instead of using globals.
* feat: If configured, adds the user's region to presence.
* fix: Guards against accessing undefined properties,
and uses the crossRegion variable from the config.
* style: Fixes formatting.
* ref(JSPC): simplify SSRC owner in P2P
There's no need for any extra extensions for SSRCs owner signalling in
P2P, because it's always the remote peer who owns them.
This also fixes a problem where no SSRC owner was added for 'source-add'
in P2P (for JVB conference Jicofo adds that).
* fix(TPC): always advertise 'sendrecv'
Our media direction is only ever updated on the remote side with
the initial offer (or answer). Because of that we want to advertise
'sendrecv' even if we start with no video (or audio) track.
It is OK to adjust this direction in the localDescription getter,
because it's adjusted again in the setter to the correct value based on
local tracks, so the SDP transformation chain still works fine.
feat(JingleSessionPC): add options for P2P evaluation
Add 'forceJVB121Ratio' config option which allows to enforce JVB
conference even if the P2P connection can be established. In such case
'forceJVB121' permanent analytics property will be added.
Will also set 'p2pFailed' analytics property to 'true' in case ICE fails
on the P2P connection.
Reorders JitsiTrackEvents.TRACK_AUDIO_LEVEL_CHANGED event arguments by
putting TraceablePeerConnection at the end. This way it's easier to
treat it as "library internal".
Will actively terminate P2P session by the responder (not moderator) in
order to shutdown P2P in case of eventual initiator's crash. Otherwise
the responder will stay in P2P for too long (until P2P ICE fails).
Prevents from printing Jingle 'session-terminate' error response in case
both responder and initiator terminates their sessions simultaneously
(gracefully). In that case 'item-not-found' error is returned by each
party, because the session is removed immediately from the memory on
termination (see strophe.jingle.js).
fix(JitsiConference): case for stopping JVB transfer
If for any reason invite for the JVB JingleSession is delayed and
arrives after the P2P connection has been established then
the media transfer needs to be disabled after the offer is accepted.
If the app depends on tracking current tracks state using
"track added/removed" events, the tracks will be leaking if
JitsiConference.leave() method is used. That's because peerconnections
are closed and removed from RTC module and then from onMemberLeft those
events will not be fired, because tracks will be gone with
the peerconnections.
Also removes error message log, because it no longer makes sense if
the tracks can be removed early by "stop P2P" logic without actually
removing them from the TPC. Then when the TPC is closed it will try to
emit the events again. But this time it will not match any tracks in
the JitsiParticipant, because it has been removed already.
Avoid a crash if for whatever reason there is no JVB JingleSessionPC at
the time when P2P is being stopped.
Add FIXME about possible situation where Jicofo invite will arrive after
the P2P has been established already.
Dominant speaker detection which is just based on current audio level of local or remote p2p track. The threshold value is the same used for talk while muted detection.
* move all local deployment properties into window.jitsiAnalyticsPermanentProperties property
no longer need to set jitsiRegionInfo from external_connect, now set from Jitsi meet local.html, can be customized by deployment
* change to using shorter lines by extracting longer property name into a shorter local variable name
* changed to using more generic variable name jitsiDeploymentInfo after discussion with the team
added comment describing source of this variable
* doc(JitsiConference): deprecate 'isInLastN'
The 'isInLastN' method should not be used for the UI purposes, but
ParticipantConnectionStatus value should be used instead.
* fix(ParticipantConn..Status): speed up INACTIVE transition
Before this change when user's video stops playing, after user is
removed from last N we were waiting 2 seconds, before going to INACTIVE
state. This commit reduces the time to 500ms for such case.
* fix(ParticipantConn...): reduce logging
Reduce logging verbosity.
* fix(ParticipantConn...): handle LastN == 0
When LastN is set to 0 we should not rely on video playback and last N
set for figuring out participant connection status.
* fix(JitsiConference): undefined participants
Fixes a crash when this.participants field is accessed from _init.
feat: Allow override of the infrastructure level channel LastN value.
For debugging purposes, it's sometimes useful to override the
infrastructure level channel LastN value via the URL (i.e. by launching
https://meet.example.org/room#config.channelLastN=X). This commit
achieves this by calling the JitsiConference.setLastN method during the
Jitsi conference initialization phase.
* feat: multiple, simultaneous RTP stats
Makes it possible to have remote RTP stats running for more than one
peerconnection at a time.
* feat(stats): report RTT all the time
Will report JVB RTT (and end to end) while in P2P mode and vice versa.
* fix(JitsiConferenceEvents): remove CONNECTION_STATS
CONNECTION_STATS event is no longer emitted.
* fix(AvgRTPStatsReported): users with no video
Do not include FPS == 0 in average remote FPS calculation. Report NaN
for local FPS when video muted or no video device. NaN will be reported
for avg remote FPS if no video is received.
* fix(AvgRTPStatsReported): reset total packet loss
* feat(AvgRTPStatsReported): report 'screen' FPS
Will report average FPS for screen videos separately from camera videos,
but only if available (camera video reports NaN FPS when not available).
* fix(AvgRTPStatsReported): end2endRTT
Needs to report JSON with value.
* feat(AVG RTP stats): separate audio and video bitrate
Will report average audio and video bitrates separately.
* doc(JitsiConference): try to improve comment
* fix(AvgRTPStatsReporter): remove confusing reset
There's no a clear reason for doing reset there.
* ref(AvgRTPStatsReporter): rename var
AvgRTPStatsReporter will calculate arithmetic means of 'n' samples
and submit the values to the analytics module. The 'n' value is
configurable through 'avgRtpStatsN' conference config option. When set
to non-positive value the AvgRTPStatsReporter will be disabled.
The following values are reported:
- average upload bitrate => 'stat.avg.bitrate.upload'
- average download bitrate => 'stat.avg.bitrate.download'
- average upload bandwidth => 'stat.avg.bandwidth.upload'
- average download bandwidth => 'stat.avg.bandwidth.download'
- average total packet loss => 'stat.avg.packetloss.total'
- average upload packet loss => 'stat.avg.packetloss.upload'
- average download packet loss => 'stat.avg.packetloss.download'
- average FPS for remote videos => 'stat.avg.framerate.remote'
- average FPS for local video => 'stat.avg.framerate.local'
- average connection quality as defined by
the ConnectionQuality module => 'stat.avg.cq'
If the conference runs in P2P mode 'p2p.' prefix will be added to
the event's name. Any pending calculations are wiped out on every switch
between P2P and JVB modes and samples have to be collected from
the start.
* ref(CallStats): cleanup constructor
Changes CallStats constructor to not take the whole JingleSessionPC as
it only needs an alias and the TraceablePeerConnection instance.
Describes the arguments in JSDoc.
* ref(CallStats): rename var
Everything is callstats c'mon...
* ref(CallStats): remove _checkInitialize
The _checkInitialize was trying to workaround CallStats lib issue
without really checking for any specific type of error on whether or not
it makes sense to retry.
Also it depended on some internal fields of 'callStatsBackend' and was
binding 'initCallback' to the backend instead of CallStats instance
which made no sense (it means it's very likely this functionality was
broken anyway).
It would be hard to fix it in a clean way, because CallStats instance
fields would have to be stored in static variables in order to make
the initCallback work (called from '_checkInitialize').
We also need to have more than one CallStats instance running at the
same time, because of the P2P which makes things even more complex.
* fix(CallStats): do not catch 'sendApplicationLog'
Wrapping 'sendApplicationLog' in 'tryCatch' will result in an endless
loop, because it will be logged on the logger.error which then tries
to send the logs immediately again.
* ref(CallStats): do not use call on static
There's nothing more confusing that seeing 'this' in a static method.
Wow maybe these methods are not really static !?
* ref(stats): fix var name
* feat(stats): report P2P to CallStats
Will create new CallStats fabric for the P2P peerconnection in order to
log peer to peer connections.
Refactoring was required in the statistics and CallStats module to be
able to have more than one CallStats instance. Because each CallStats
fabric reports one peer connection now each CallStats will correspond to
one TraceablePeerConnection. CallStats instances are now stored in a Map
mapped by the TraceablePeerConnection.id field.
In order to be able to execute some global/static CallStats reporting
methods all Statistics instances also need to be stored in a static
field.
CallStats API backend(new callstats()) will be initialized only once for
the values provided in the first call to initializeBackend. It is not
possible to have more that one CallStats backend running at the same
time (at the time of this writing). If we would have a routine for
disposing global "Statistics" module we could try to cleanup static
reference and allow to initialize it with new values (but no such use
case yet).
* ref(CallStats): move initCallback
Since there is no alphabetic order preserved in this file anyway at
least place it closer to it's usage.
* ref(CallStats): remove tryCatch
Temporarily removes tryCatch to make the ES6 conversion easier.
* ref(CallStats): convert to ES6
* style(CallStats): fix indentation
* fix(statistics): use import for CallStats
* ref(CallStats): convert static methods
Some of the methods should not be static, because it only make sense
to call it when there is CallStats instance available.
* ref(CallStats): rename var
* doc(CallStats): remove misplaced comment
* chore(CallStats): remove invalid eslint comment
* fix(CallStats): undefined CallStats namespace
If no CallStats ID namespace option is provided the conference will be
reported without it.
* style(stats|CallStats): remove extra lines
* fix(CallStats): do not log error from tryCatch
GlobalOnErrorHandler calls logger.error anyway.
* fix(CallStats): cleanup tryCatch
If I understand correctly our initial intention with doing tryCatch was
to avoid crashing the whole app in case the CallStats backend would
crash. With this commit the tryCatch is done by wrapping original
backend instance methods or using explicit try catch block where
the method is called only from one place or a value needs to be returned
in case of a crash.
* ref(CallStats): make backend a static var
* fix(CallStats): invalid eslint comments
* ref(CallStats): use for..of
* ref(CallStats): fixes around REST args
* ref(CallStats): rename var
* ref(CallStats): reorder static methods
Also renamed some callbacks
* doc(CallStats): adds some docs
* ref(CallStats): make methods not static
Both 'sendDominantSpeakerEvent' and 'sendScreenSharingEvent' methods are
not really static as they require instance to be called.
* fix(CallStats): invalid key
* fix(CallStats): reduce amount of debug logs
* feat(p2p|CallStats): log hold/resume
Will put CallStats fabric for the JVB connection "on hold" while in p2p
mode.
* doc(CallStats): add FIXME
* doc(JitsiConference,CallStats): typos and renaming
When muted track was added to conference "mute" operation was
executed again which was not executing anything because the state
of the track was already muted.
damencho
bgrozev
hristoterezov
Lyubo Marinov
paweldomas
Saúl Ibarra Corretgé
virtuacoplenny
George Politis
Saúl Ibarra Corretgé
Iñaki Baz Castillo
Aaron van Meerten