In peer.js offer generation method, the provided constraitns are only
being type checked. It needs a value check when the constraint is a
boolean.
This commit should prevent useless transceivers from being added when
explictly specifying audio/video as `false`
There's an edge case in finnicky networks where ALG-like firewalls
tamper with USE-CANDIDATE STUN packets and, consequently, bork ICE-lite
connectivity establishment. The odd part is that client-side gathering
seems to complete if intermediate STUN bindings work (before the final
USE-CANDIDATE), which may cause the peer not to generate relay
candidates == connectivity fails.
This adds the `public.kurento.gatheringTimeout` option to forcefully extend
the candidate gathering window in peers that act as offerers. The
behavior is as follows: if the flag is set (ms), the peer will wait
either the gathering completed stage or, _at most_,
public.kurento.gatheringTimeout ms before proceeding with calls chained
to setLocalDescription.
This option is disabled by default and intentionally ommited from the
base settings.yml file as to not encourage its use. Don't use it unless
you know what you're doing :).
There are still a bunch of edge cases and issues with reconnection
scenarios for video:
- Signaling socket refuses to reconnect once maxRetries expire
- Race conditions on local stream attachment: local camera wouldn't be
correctly rendered _if_ the attached stream existed _without_ video
tracks yet
- Video tracks leak on local streams when replacing them (virtual bgs)
- Completely ignoring Meteor state when trying to reconnect cameras
- Streams aren't proactively stopped when the signaling socket dies
- Outbound request queues aren't isolated by stream nor are they
flushed when a newer peer with the same ID is created
- Server originated negotiation errors won't trigger a local peer
cleanup - thus leaving dangling peers that take way too long to
reconnect
This commit fixes or improves all of the aforementioned issues, +:
- Remove unused arguments in the peer (client->SFU) 'start' request
- Prevent crashes when trying to render video-list-items without user
data (which might happen on re-connections)
RTCRTPSender exposes DSCP marking via `networkPriority` in the encodings
configuration dictionaries. That should allow us to control
QoS priorities for different media streams, eg audio with higher network
priority than video. The only browser that implements that right
now is Chromium.
To use this, the public.app.media.networkPriorities configuration in
settings.yml. Audio, camera and screenshare priorities can be controlled
separately. For further info on the possible values, see:
- https://www.w3.org/TR/webrtc-priority/
- https://datatracker.ietf.org/doc/html/rfc8837#section-5
kurento-utils is unmaintained. It's served us well, but its age
shows. We need to transition to something else if we want to
have better maintainability and include simulcast, multistream, ...
This introduces a simplified/leaner wrapper kit that's almost
API-compatible with what we use right now - so widespread changes
are minimal). It's easier to maintain/read/transition from. This
can be read as an intermediate step to transitioning to
something definitive (ie mediasoup-client).