* Metrics for participant active, i. e. fully established.
- Egress stub for v2 API
- Fix the participant canceled counter 🤦
- Add active counter -> this is increment when a participant becomes
active, i. e. primary peer connection established. Can be used to
monitor node wise connection establishment issues.
- Add singnalling validation fail counter.
With this, we have
- signalling validation fail
- signalling failed --> this is when the `startSession` fails
- signalling connected -> signalling is succesful and can send back
joinResponse to client
on media connection side
- rtc_init -> start
- rtc_connected -> participant session created (joined)
- rtc_active -> primay peer connection established
- rtc_canceled -> could not proceed with RTC connection due to not being
able to resume.
* signalling counters deps
* revert pion/webrtc to 4.2.12 to get SCTP without interleaving
* go back to pion/webrtc 4.2.11 and sctp 1.9.5
* telemetry: split webhook-processed hook registration out of NewTelemetryService
NewTelemetryService used to register a notifier processed-hook on the inner
*telemetryService directly. That made it impossible for downstream wrappers
(e.g. cloud's TelemetryService that overrides Webhook to fan out to a v3
observability pipeline) to intercept webhook events without double-firing
the legacy emission.
Lift the registration into a new exported helper RegisterWebhookHook, and
have the standalone server's wire provider createTelemetryService call it
right after construction so behavior is unchanged for callers that don't
wrap the service.
When a client hits /rtc/v[01]/validate with a base64 WrappedJoinRequest
whose embedded JoinRequest.ClientInfo is unset, validateInternal called
AugmentClientInfo with a nil *ClientInfo and panicked at ci.Address =
GetClientIP(req). The non-wrapped branch already allocates via
ParseClientInfo; do the same here so pi.Client always gets at least the
resolved client Address.
Sfu will fallback to retransmit packet by media stream ssrc if rtx
is not negotiated (client doesn't have), so we should not disable
rtx explicitly (by codec config).
Fix#4519
* Apply ttl check only when authenticate allocation creating
TTL check could reject allocation/persmission refresh in
security enhancement #4505, cause long-live session disconnect
when turn credential is expired.
Only check ttl on allocation creating to prevent abusing leaked
credential but keep long-live session work.
This patch updates the check for auto creating rooms to also
consider the RoomCreate grant per token instead of just the
global config option.
With this patch, applications can decide on their own whether
users or which users can auto create rooms. This allows
applications that rely on auto creation (saving an API call)
to co-exist with those who might want to mint tokens for
subscribe-only users.
Specifically LaSuite Meet relies on the auto create behavior,
however enabling the global config option would make a
MatrixRTC deployment vulnerable to abuse, as users on remote
homeservers get tokens in order to subscribe.
* Add TURN permission handler.
- Turn off permissions to private/link local/multicast and internal IPs
- Add a list of CIDRs that can be used for more things to deny
permission to.
* unused
* add config for allowing private IPs, used in testing
* add a TTL to user name and use it to auth
* allow list for restricted peer CIDRs
`iceServersForParticipant` builds UDP TURN URLs by interpolating the
node IP directly into a format string:
fmt.Sprintf("turn:%s:%d?transport=udp", ip, port)
When `NodeIP.V6` is set, `ToStringSlice()` includes the bare IPv6
address, producing URLs like:
turn:2a05:d014:ee4:1201:7039:38c:f652:a252:443?transport=udp
RFC 3986 §3.2.2 requires IPv6 addresses in URIs to be enclosed in
square brackets. Without them the port is ambiguous and WebRTC clients
(e.g. libdatachannel) reject the URL with "Invalid ICE server port".
Use `net.JoinHostPort` which handles bracketing for IPv6 and is a
no-op for IPv4, producing well-formed URLs:
turn:[2a05:d014:ee4:1201:7039:38c:f652:a252]:443?transport=udp
turn:1.2.3.4:443?transport=udp
* fix: ensure num_participants is accurate in webhook events (#4265)
Three fixes for stale/incorrect num_participants in webhook payloads:
1. Move participant map insertion before MarkDirty in join path so
updateProto() counts the new participant.
2. Use fresh room.ToProto() for participant_joined webhook instead of
a stale snapshot captured at session start.
3. Remove direct NumParticipants-- in leave path (inconsistent with
updateProto's IsDependent check), force immediate proto update,
and wait for completion before triggering onClose callbacks.
* fix: use ToProtoConsistent for webhook events instead of forcing immediate updates
* Log join duration.
Also revert the "unresolved" init. Defeated the purpose of log resolver
as it was resolving with those values even if not forced. Instead set it
to "unresolved" if not set when forced.
Join duration is not reset if resolver is reset as that happens on
moving a participant and there is no new join duration in that case.
* explode
Some e2e is failing due to subscriptions happening late and the expected
order of m-lines is different. Not a hard failure, but logging more to
make seeing this easie.
Subscription can switch between remote track and local track or
vice-versa. When that happens, closing the subscribed track of one or
the other asynchronously means the re-subscribe could race with
subscribed track closing.
Keeping the case of `isExpectedToResume` sync to prevent the race.
Would be good to support multiple subscribed tracks per subscription.
So, when subscribed track closes, subscription manager can check and
close the correct subscribed track. But, it gets complex to clearly
determine if a subccription is pending or not and other events. So,
keeping it sync.
This allows for abstracting away how the stop is implemented - default implementation stays the same - the existing OSS egress launcher just calls the existing Stop method on the client.
Raja Subramanian