* fix(RTC): Adjust the media direction for p2p conn. For p2p connections, the media direction needs to be adjusted after every source-add/source-remove is processed based on the availability of local sources.
* fix(RTC): Use a enum for media direction.
5738c80baf...d9d9b7fc31
* fix(JingleSessionPC): Disable unified-plan for p2p. Disable cross browser p2p using unified plan until all the issues are fixed.
0993c8e93d...5738c80baf
* fix(LocalSdpMunger): Fix unit test.
* fix(CodecSelection): Call RTCRtpTransceiver#setCodecPreferences before renegotiation. Call RTCRtpTransceiver#setCodecPreferences with the preferrred codec order before every createOffer/createAnswer. This ensures that the codec preference is enforced even when there is no local description available yet while the preferred codec is being set immediately after media session creation.
* fix(JingleSessionPC): Add a workaround for chrome issue. The 'signalingstatechange' event for 'stable' is fired after the 'iceconnectionstatechange' event for 'completed' is fired on chrome in Unified plan. This prevents the client from switching the media connection to the p2p connection once the ice connection for p2p gets established.
* fix(Logging): Log enhancements. Add a preifx to logs for idenitifying the type of TPC/jingleSessionPC.
* feat(TPC): Enable unified-plan support for Chromium based browsers. This can be controlled through the config.js option 'enableUnifiedOnChrome'.
* fix(TPC): Do not configure encodings on Safari until reneg. Avoid configuring the encodings on Chromium/Safari until simulcast is configured for the newly added track using SDP munging which happens during the renegotiation.
* fix(TPC): Do not configure encodings on chromium immediately after replace track. Avoid configuring the encodings on chromium immediately after replace track since the encoding params are read-only until the renegotation is done.
* fix: send json message (#1180)
be3e2a69f2...3fb44f7695
* fix(SDP): Add missing msid for p2p sources.
* fix(TPC): Don't convert plan-b<->unified-plan SDPs for p2p.
* squash: Implement review comments.
* fix(JingleSessionPC): Do not try to re-use inactive mid for new remote ssrcs. The direction was marked as 'inactive' only on Firefox as Safari had audio issues when an inactive mid is re-used. Chrome (in unified-plan) needs the direction of the mid in remote desc to be set to 'inactive' for a 'removetrack' to be fired on the associated media stream whenever a remote source is removed.
* fix(SDP): Drop SSRCs whenever the transceiver direction is 'inactive' or 'recvonly'. This is needed only for JVB connections. Add unit tests for LocalSdpMunger.
* fix: Ignore startAudioMuted/startVideoMuted for p2p. The tracks will not be added when the call switches from jvb to p2p for an endpoint that joins muted by focus.
* fix(RTC): Do not suppress the source updates on Firefox. If the msid attribute is missing, then remove the ssrc from the transformed description so that a source-remove is signaled to Jicofo. This happens when the direction of the transceiver (or m-line) is set to 'inactive' or 'recvonly' on Firefox. Not signaling these source updates creates issues with remote track handling on the other endpoints in the call.
* fix(RTC): Set transceiver direction after RTCRtpSender#replaceTrack. This fixes the issue where TRACK_REMOVED event is not fired when a remote track is removed from the peerconnection. Fixes https://github.com/jitsi/lib-jitsi-meet/issues/1612 and https://github.com/jitsi/jitsi-meet/issues/8482.
60c5667957...be3e2a69f2
* fix(caps): Disable TCC on Firefox. There is a known issue with Firefox where the BWE gets halved on every renegotiation causing the low upload bitrates from the Firefox clients.
* fix: Drops unused config, fixesjitsi/lib-jitsi-meet#1620.
* fix(e2ee): destroys olm session on disabling e2ee
f95a455c08...60c5667957
* fix(TPC): Return default codec if the local sdp is not available. Get the correct media type when generating the source identifier.
88560a8a5e...9eb4af1e80
* feat(prosody-modules): Moves a function for getting room to util.
* feat: Audio/Video moderation.
* squash: Fix docs.
* squash: Changes a field name in the message for adding jid to whitelist.
* squash: Moves to boolean from boolean string.
* squash: Only moderators get whitelist on join.
* squash: Check whether in room and moderator.
* squash: Send to participants only message about approval.
Skips sending the whole list.
* feat: Separates enable/disable by media type.
Adds actor to the messages to inform who enabled it.
* squash: Fixes reporting disable of the feature.
* squash: Fixes init of av_moderation_actors.
* squash: Fixes av_moderation_actor jid to be room jid.
* squash: Fixes comments.
* squash: Fixes warning about shadowing definition.
* squash: Updates ljm.
* fix: Fixes auto-granting from jicofo.
* squash: Further simplify...
* fix(JingleSession): Move the ssrc identifier generation to LocalSdpMunger.
* fix(logger): Logging enhancements. Get rid of noisy logs related to SDP transformations which are redundant. Fix formatting and add missing information.
7cbd9c8f2a...923aa449c4
* fix(quality-control): Propagate the height constraints to p2p session. If the application is using the new receiver constraints, propagate the height constraint to the p2p session as well.
* build(deps): bump lodash from 4.17.19 to 4.17.21
* chore(deps): bump hosted-git-info from 2.8.8 to 2.8.9
74a90f7035...7cbd9c8f2a
* fix(quality-control): fix constraints sent on channel initialization. Do not send old format constraints if no constraints are set before the channel is initialized.
* chore(deps) run npm audit fix
* chore(deps) update webrtc-adater@8.0.0
86c7a35817...74a90f7035
* Add dependency for promise.allSettled. Older chrome versions like M72 do not support Promise.allSettled.
* fix(conference): Enable p2p for unified plan clients.
* fix(TPC): Use addTrack instead of addStream in Unified-plan impl.
* Add missing spaces in debug logs.
ad5692d6aa...e362c89eb6
* fix(SDP): Move all SDP related files to a different dir. SDP utility classes are spread across RTC and XMPP directories now, moving these class files to a 'sdp' directory.
* fix(stats): Return promise for getStats. Switch to returning a Promise for getStats. Reset frame rate stat to 0 when video is suspended as a result of endpoint falling out of last-n.
* Fix: sysMessageHandler not deleted (#1590)
* task(e2ee): switch back to GCM
463e213b3f...7667117117
* fix(quality-control): Send the new constraint on join. Fixes the case where the old format height constraint is sent on join for a jvb media session.
7dedb59b9c...463e213b3f
* fix(quality-control): Switch to new receiver constraints by default. Use the new receiver constraints unless it is explicitly disabled through config.js.
3c9913ed61...7dedb59b9c
* fix(JingleSession): Increase the ICE candidate gathering timeout to 150ms. This will reduce the numbers of transport-info IQs sent by the client.
* fix(TPC): Fix error handling for getStats.
ca325f5ef9...0dc1540a44
* fix(stats): Use promise-based getStats on all browsers. Get rid of the browser specific keys and use the standard spec-compliant fields for stats. Get the resolution/fps for remote streams from 'inbound-rtp' stats. Use the 'track' stats for the local resolution/fps since these take the active simulcast streams into account.
8b3dc59374...ca325f5ef9
* fix(SS): Implement a 2500Kbps limit for VP9 SS.
* fix(RTC): Remove stream effect before disposing the track. Remove the effect instead of stopping it so that the original stream is restored on both the local track and on the peerconnection. Fixes issues when a stream with effect applied is replaced on the pc after it is muted, also fixes https://github.com/jitsi/lib-jitsi-meet/issues/1537.
* fix: Drops unused config.
1f3f85978d...baa78aca40
Steffen Kolmer
Jaya Allamsetty
damencho
Avram Tudor
Saúl Ibarra Corretgé
Tudor-Ovidiu Avram
Calinteodor
dependabot[bot]
Vlad Piersec
titus.moldovan