6bd7fac875
46c43095 fix goreleaser workflow, version 1.13.1 (#4577) e0815be2 chore: improve docker test shutdown reliability (#4576) bfd9deff expose TCPFallbackRTTThreshold and AllowUDPUnstableFallback via config (#4556) b93c1e16 Release v1.13.0. (#4573) fd452212 Update mediatransportutil to get ICE candidate timeout config (#4572) 8be8c74a Update github workflows (#4463) c4e41872 Update go deps to v1.17.2 (#4462) dc8e0310 Update go deps to v4 (#4482) 20fd1ad2 turn: allow for providing secret via file (#4564) 6590570d Pin pion/dtls to v3.1.2 (#4570) cdbbee1f deps: bump protocol + psrpc to latest tips (#4565) d290de81 Correct config comment (#4563) 77ecf920 rtc: report participant session end time on room move (#4561) 63be96f6 Prevent panic from nil(illegal) syncState.Subscriptions message (#4560) 835ef1b3 Metrics for participant active, i. e. fully established. (#4557) 5bd42534 Document of advertise_internal_ip and external_ip_only (#4554) 356ae211 Config documentation for advertise_internal_ip and skip_external_ip_validation (#4552) 7c319a67 rtc: prevent duration reporting for inactive participants (#4550) 2dd5e632 telemetry: split webhook-processed hook out of NewTelemetryService (#4548) 222177a9 service: prevent nil deref in validate with wrapped join request (#4547) dd7580b4 Protect against nil clientInfo (#4546) 145689e6 Start tracking Twirp method request latency in prometheus too, not just in logs (#4545) cde89627 rtc: emit per-data-track bytes via BytesTrackStats (#4540) 2e22911d Remove backwards compatibility support for TURN auth. (#4539) 062d1219 Use NACKQuueInterface type. (#4538) 7f08b04c Add IsIntentionalDisconnect helper (#4537) 1ab2bf04 Clean up packet size logging (#4536) 8ab92a80 Don't require media sections when joining (#4535) 019a6640 rtc: report participant kind code and details (#4534) 77595d38 TEL-336: fix sip error categorization (#4528) f303f499 Always enable rtx codec (#4533) e4a8a55c Check Less and LessEq in version compare. (#4532) 37eb7a32 Release v1.12.0 (#4529) 4a7b1e85 Create NACK tracker only once. (#4527) 89faaeba Apply ttl check only when authenticate allocation creating (#4526) b32933b0 Log details of RTCP packets. (#4525) 8b79ec9e Support SIP auth realm for inbound. (#4522) 4b8db3cf Add integration test for TURN auth failures (#4524) ef2e5efe Log large packets receive/send. (#4521) d1236750 feat: auto create rooms for tokens with the RoomCreate grant (#4320) 7a3e595b apply room tags from JWT grant room configuration (#4518) ab7fdeab add AssignmentHook to AssignJob; propagate websocket write errors (#4516) cf20c9cd Add expiry to TURN password. (#4515) 20d4a3a1 Populate data track loggers with context (#4514) 12fff29a allow setting agent job assignment url (#4512) ba366fc7 Fix SIP media config upgrade. (#4511) 8fbc5adf update protocol for protojson (#4510) 3de6f517 Add TURN permission handler. (#4505) 8ffcef93 Update protocol to support SIP media config. (#4509) 1ab1e072 test: verify upstream and downstream connection stats end-to-end (#4508) c4fd71a5 Fix sense check in DeltaInfo gathering (#4507) 803999ef rename agent environment to deployment (#4506) bacc21e6 add helper to check for agent worker endpoint (#4503) 253f977d add duration seconds reporting (#4500) ffab3bd3 add agent environment (#4498) ccdf23c8 Use mediatransportutil/codec package, no functional change (#4497) 680703f2 Include reception reoprts in receiver report callback. (#4496) f51798bc Fix publish-only limitations being incorrectly applied to receivers (#4495) a002337d Legacy TrackInfo.Simulcast flag. (#4493) af1dcc88 Add CloseWithReason to agent SignalConn interface (#4492) d7c2daf1 report all simulcast layers (#4491) c1ad2b22 Misc optimisations. (#4490) 19b9e8c0 Additional data tracks logging (#4489) 743d9c8b add support for client capabilities (#4461) fc47e478 Close peer connection unconditionally to unblock set local/remote (#4485) 639406eb Update module github.com/pion/ice/v4 to v4.2.3 (#4481) dc6b7505 reduce some heap use in packet path (#4478) f3b80b28 fix: wrap IPv6 addresses in brackets in UDP TURN URLs (RFC 3986) (#4476) 3a7f2628 Turn off transceiver re-use on Safari. (#4474) d84f3d7a add more types to signum (#4473) 701a37c2 Convert sort.Slice -> slices.SortFunc (#4472) 85be9d70 Avoid stream allocator event data cast to interface and back. (#4471) b43685e8 Keep a shadow copy of tracks for use by different stream allocator state (#4470) 27c2b149 Consolidate RTCP packets and do RTCP callback outside lock. (#4469) 31083307 do not log data track stats if not started (#4468) 9ee06635 feat(pion/ice): replace deprecated NAT1To1 with SetAddressRewriteRules (#4466) 8ccad68d Release v1.11.0 (#4459) dbf5cf61 Store concrete ICE candidate for remote candidates. (#4458) 2a04bc3c fix publisher frame count reporting for simulcast streams (#4457) 1d804737 fix: limit join request and WHIP request body to http.DefaultMaxHeaderBytes (#4450) 3cfb71e7 Use Muted in TrackInfo to propagated published track muted. (#4453) 69aa9479 Some drive-by clean up (#4452) 6c81f678 Add subscriber stream start event notification (#4449) ce1bf47b Revert "fix: ensure num_participants is accurate in webhook events (#4265) (#…" (#4448) cdb0769c fix: ensure num_participants is accurate in webhook events (#4265) (#4422) c91e79af Switch to stdlib maps, slices (#4445) ea7b9c6f Update module github.com/livekit/protocol to v1.45.3 (#4435) 97378368 Update go deps (major) (#3179) d6aef547 Update go deps (#3862) afc9feae Update github workflows (#4331) 4b385612 chore: pin GH commits and switch to golangci-lint (#4444) 2974ba87 Unsubscribe from data track on close (#4443) 5dc4e90d Apply IPFilter when get local ip (#4440) 88c77dc6 compute agent dispatch affinity from target load (#4442) 8fe99377 Log join duration. (#4433) 0a503a57 Add `Close` method for UpDataTrackManager and call it on participant (#4432) 55912dff Add some simple data track stats (#4431) 050909e6 Enable data tracks by default. (#4429) 72c7e65c chore: log API key during worker registration (#4428) 8a67dd1b Do not close publisher peer connection to aid migration. (#4427) 91e90c10 Add some more logging around migration. (#4426) c6ddc879 isExpectedToResume is based on whether flushing or not. (#4425) 7d06cfca Keep subscription synchronous when publisher is expected to resume. (#4424) 934f8598 Clean up data track observers on unsubscribe. (#4421) 9674ac48 Cleaning up some logs and standardising log frequency. (#4420) 7b925304 Drop time inverted packets in RED -> Opus conversion. (#4418) 4d8d232a ensure participant init is correctly serialized for logging (#4417) 4fe80877 Log time inversion between incoming packets (#4415) 248d7394 Guard against timestamp inversion in RED -> Opus conversion. (#4414) 9ab8c1d5 clear track notifier observers on subscription teardown (#4413) 397cd09a Embedded turn test (#4412) 56326654 Prepare release v1.10.1 (#4408) e9b113c8 Make the TURN bind address configurable and allow for multiple addresses. (#4315) 4bc5e6bb Address malformed H264/H265 parsing issues. (#4407) 77a0a4fc AV1 parser overflow fix. (#4405) ff7fd7ed feat(agent-dispatch): add job restart policy (#4401) 34bd1e08 do not log roll over for padding only packets (#4396) 13d02ee9 add deadline to dtls connect context (#4395) 9055a349 Path check helpers (#4392) 1f1eeb68 Fallback to servicestore if rpc is unavailable (#4391) 59e9bb41 Fix TURN server URL (#4389) 9e0a7e54 Close both peer connections to aid migration. (#4382) 9474c807 route participant reads through PSRPC instead of Redis (#4387) a5333a86 add packet trailer stripping support (#4361) 0e3d765d Release v1.10.0 (#4380) bc3aeaf3 Update grpc to address CVE-2026-33186 (#4381) 8cdd6f4c Replace deprecated io/ioutil with io in whipservice (#4375) 89410df7 handle AGENT_ERROR disconnect reason (#4339) 8f984c77 Fix repair stream ID reporting for RTX pairing. (#4369) cdfaacfc Restart nacker on OOB sequence number restart. (#4368) 4a9e0045 Update protocol. (#4367) 750d5904 Add API to restart lite stats. (#4366) c8bb2578 Rename log field "pID" to "participantID" for consistency (#4365) 77fc74a7 Do not block all ext ID determination on stream allocator listener (#4364) 90a46fab Do not kick off migration of closed participant (#4363) 5dc2e7b1 Switch data track extension to 1-byte ID/length. (#4362) 7323ad02 Sample data send error logging. (#4358) 0d34e455 Add option to not re-use transceiver in e2ee. (#4356) 95225ff2 don't require media section for dual peerconnection mode (#4354) e9639538 Refine ipv6 support (#4352) b34b0472 Add StopEgress function to the EgressLauncher interface (#4353) 69235ed2 update readme (#4340) db1a8046 defensive check for peer connection instance (#4350) cb7dc2d0 TEL-405: support originating calls from custom domains (#4349) 7eaaaada Mark last run of grow bucket outside goroutine. (#4348) caa47522 Add option to require media sections when participant joining (#4347) 087050d1 Release v1.9.12 (#4346) 493e87df Fix SIP client timeout. (#4345) 52c28a93 Log a bit more details of out-of-order TWCC feedback report. (#4343) 516aeabf Use ParticipantTelemetryListener of LocalParticipant. (#4342) b3510565 Exclude ice restart case from offer answer id mismatch warning (#4341) 303657bc feat: make INSTALL_PATH overridable in install script (#3954) 9d418689 Send participant left event after track unpublished for moved (#4334) bab91868 do not discount packets lost on duplicate packets (#4333) 939794cf mark + restart (#4329) 75f9c462 Add debug for receiver restart. (#4328) 74891f30 Protect against incorrect temporal layer. (#4327) f51b2732 Update pion/webrtc and deps to update dtls (#4326) b81bac0e Key telemetry stats worker using combination of roomID, participantID (#4323) 2d40449f Update self-hosting deployment documentation link (#4312) 03e90dd7 Fix for some CodeQL reported issues (#4314) 77c858f0 Log stats worker when it not closable. (#4313) a6035212 ESP32 client info (#4267) 478e486a Copilot suggested improvement to Github Actions permissions (#4310) cbd2f82d Copilot suggested improvement to Github Actions permissions (#4311) a9b8d40d Publish is always on publisher peer connection. (#4307) 8ae56406 generate & log egressID for start egress request (#4303) 35d7ef88 Avoid alloc in RTPStatsReceiver.Update (#4302) bb744916 More optimisation in RTPStats module. (#4298) cefd5da9 Optimise some bits in rtpstats_receiver (#4297) 52a4b848 Read client protocol from query param (#4294) 195b17f6 Populate client_protocol field in ParticipantInfo (#4293) 370e0a4d Set up audio config in audio level module when config is updated. (#4290) f3e9b688 Do not increase max expected layer on track info update. (#4285) a9849340 Avoid logger data race. (#4284) 97016627 Do not hold lock when creating buffer (#4283) 6b68e3d5 Create buffer if needed when a PLI is requested. (#4282) 3cca7180 use separate allocation for signal stats telemetry guard (#4281) d2bae34d refresh telemetry guard on participant move (#4280) 333f0349 clear reference guard when resetting signal stats (#4279) d1bab17b Add session duration and participant kind to closing log. (#4277) 1e689e1a Reducing some info level logs. (#4274) 88facc02 adds a test to ensure agent worker errors cause disconnection (#4273) 76a41a7a Generate config flags (#4268) 700e1788 require participant broadcast when metadata/attributes are set in token (#4266) b61799ec Ignore parse addr error when add remote candidate (#4264) 01bd966f Add silent frame for pcmu/a (#4258) 0c33b8c6 chore: move codecs/mime stuff to protocol (#4255) 165c1735 Update livekit protocol to v1.44.0 (#4254) 343f12b8 Wrapping SIP errors for invalid argument and not found (#4253) aea044c5 Defer setting clock rate in RTPStats module till codec is bound. (#4250) d9f716c1 FIx receiver restart race (#4248) 0508ee9f remove copy/paste left over line (#4246) 40408407 Release v1.9.11 (#4245) f8675507 Do not remove participant from cache on disconnect. (#4241) a35a6ae7 Add participant option for data track auto-subscribe. (#4240) 07572511 Wrapping the invalid request errors for CreateSipParticipant (#4239) 843d8c3e Update Pion transport package. (#4237) a05690d2 Changing field naming of data track packet (#4235) c4015008 Clear participant version cache on disconnect (#4234) 18db4ec1 Log modified timeout of API context. (#4232) ac20ccda Log timeout in API. (#4231) 541a7d6c Change some logs to debugw (#4229) 52ab3374 Return on SDP fragment read error. (#4228) 7fae5ac9 Do not restart receiver on codec change mid-session. (#4225) dafdc36e Add option to force simuclast codec. (#4226) 0a7dd40b Use only layer 0 for SVC codecs. (#4224) 4ec0f8f4 Support OpenTelemetry tracing. Add Jaeger support. (#4222) 80ba93fa Do NACK updates as soon as flow state is processed. (#4221) 4405afe2 Use atomic pointer and return interface from RED transformer constructors (#4220) b649c2fe Remove method not needed from REDTransformer. (#4219) f0080f35 Remove enable arrival time forwarding method. (#4217) 335f4c33 Swap result sink atomically rather than closing and setting. (#4216) 46651c19 Release v1.9.10 (#4214) 08ac4ecd Support preserving external supplied time. (#4212) f6efccce report video size from media data for whip (#4211) d92f6a79 return iceservers for whip (#4210) 1a4758ed Skip restart callback if external. (#4208) dde4fb49 configurable dependency descriptor restart (#4207) 08793bea Use active at time to check for track not bound timeout. (#4206) 3606ce54 Do not warn about track not bound if participant is not ready. (#4205) b8ddd0f9 Taking interface{} -> any modernize bits (#4204) b91cd2e4 Rework receiver restart. (#4202) bb00c864 Restart API on receiver. (#4200) 25ece1e9 Minor refactor in buffer base and audio level (#4198) 599002f8 ignore PLI requests for non-video (#4196) 2510b946 Taking a bunch of go modernize suggestions. (#4194) ed8e6afc Handle repair SSRC of simulcast tracks during migration. (#4193) c6bf7a27 Fix logging key and other clean up around stream restart. (#4192) 3cb9abb6 Update pion/webrtc to v4.2.1 (#4191) 8b0efb8c Resolve RTX pair via OnTrack also. (#4190) 381bce03 Return extended sequence number only and not packet. (#4189) 6bcbf54e Always instantiate nacker when using out-of-band sequence numbers. (#4187) e71184de Store buffer after creating it. (#4186) 7c8ea115 Refactor receiver and buffer into Base and higher layer. (#4185) 4104b827 update protocol (#4183) cd99fec2 Make new path for signalling v1.5 support. (#4180) 32cd0370 Flush the ext packets on restart/close and release packets. (#4179) 1df1316b Move OnDataTrackMessage to participant listener to support replay. (#4178) e7601251 Make data message naming a bit more consistent. (#4177) a04e566d Use published track for model access in data down track. (#4176) 47c86be1 Add support for TURN static auth secret credentials (#3796) 24559a28 chore(deps): update github workflows to v6 (#3866) 0a824386 add explicit room exists servicestore op (#4175) 39bd077d Release v1.9.9 (#4174) cbb2c617 Publish/Unpublish counter match. (#4173) fb849edc Minor clean up (#4172) c28e5e45 fix(kindToProto): update protocol (#4171) 47324abd Drop run away receiver reports. (#4170) 462ec324 prevent uint overflow setting packet not found count (#4169) 5c841b8e Some logging changes. (#4168) 2f2d0a57 skip lost sequence number ranges in getIntervalStats (#4166) 898ebe05 clean up manual roomservice log redaction (#4165) 3e417253 move delete to oss service store (#4164) 5964efbb Ensure subscribe data track handles are unique (#4162) a26c4830 Add support for RTP stream restart. (#4161) 386f0b38 fix typo in clearing index when removing track from room track manager (#4158) 0abfb251 deregister observability function when participant is closed (#4157) 97aba5e7 Consistently undo update to sequence number and timestamp when the (#4156) 2317c295 Fix panic while removing track from room track manager. (#4153) a0a28ac5 Avoid duplicate track add to room track manager. (#4152) f01008f8 Revert telemetry stats worker wait configuration. (#4151) ca4b56d2 Handle case of sequence number jump just after start. (#4150) 97099cae Configurable telemetry stats worker clean up wait. (#4148) d7db7cb3 chore: fix a large number of spelling issues (#4147) cadf2ad9 Release v1.9.8 (#4145) 498304cd defensive nil check (#4144) 20f6a497 Store ddParser in atomic.Pointer (#4143) 037cb906 release ext packet if patching fails (#4142) dd598ef2 Release ExtPacket if dependency descriptor or other parsing fails (#4141) 35b0e2bc update webrtc to 4.1.8 to pick up DTLS fingerprint check during handshake (#4140) 1c1a836c Mark RTCP buffer Write as noinline. (#4138) ea9b2177 protocol deps to get inactive file adjusted memory usage. (#4137) 64f3d1e9 switch participant callbacks to room to listener interface (#4136) c6e6c021 add debug metric for tracking references (#4134) a30c79fa Use isEnding to indicate if down track could be resumed. (#4132) 8c241ecf Fix RTCP reader leak in DownTrack. (#4131) 3eef869a Do not pause rid in SDP (#4129) 8e01e595 Release 1.9.7 (#4128) 7c1a0fab Fix concurrent map access. (#4127) 14446b1c Let participant close remove the published tracks. (#4125) fa0633aa move utils.WrapAround to mediatransportutil (#4124) f8706cd4 Update pion/ice to stop gather first on close (#4123) 7954748d Data tracks (#4089) 7158d983 log bucket growth (#4122) 04b35eb6 Release v1.9.6 (#4121) ebdcead5 Update mediatransportutil to get bucket packet size limit. (#4120) 411b09f6 Release v1.9.5 (#4119) 8dcf235a Update pion/ice - attempt to address tcp packet conn close hang (#4116) 64c65143 Update mediatransportutil (#4115) 0a2943bb Clean up bits added to debug peer connection close hang. (#4114) 9c483a69 Use released version v1.8.41 of pion/sctp (#4113) 35c79a57 Update SCTP hash, had the wrong one in previous PR (#4111) e0fbbef1 Update pion/sctp with RFC9260 revert (#4110) f3c80917 Try SCTP with read deadline to unblock abort. (#4109) bd5382da Splitting transport close timeout logs. (#4108) 6d4154b8 Update pion/ice. (#4107) a6418ae2 Log more peer conenction state on close timeout. (#4105) 06d99974 Check for cancel on unsubscription/source track going away. (#4104) 7f10e18b Record join/publish/subscribe cancellations. (#4102) 40293632 Clear stereo=1 if stereo is not enabled. (#4101) 70f6def3 Add checks for participant and sub-components close. (#4100) ffbabcc7 Switch forwarding latency log to Debugw (#4098) 27d82a72 Fix "address" typo in transport logs (addddress → address) (#4097) 37a06821 logger proto redaction. (#4090) 54cf7d46 Control latency of lossy data channel (#4088) 5175c1af Lock x/tools at 0.37.0 (#4085) d510fff1 Downgrade x/tools to be able to make a release (#4084) c3ea5890 Prepare release v1.9.4. (#4083) 3a128e61 protocol bump for SIP error mapping and validation (#4081) c3964ba2 Use sync.Pool for objects in packet path. (#4066) f8b994d4 Forwarding latency measurement tweaks. (#4080) f4929f09 Revert "Revert pion/transpor to v3.0.8 (#4073)" (#4074) a04d9c48 Revert pion/transpor to v3.0.8 (#4073) 2d5054ad kind details for connector (#4072) a272e28a Log raeson for subscriber not being to determine codec. (#4071) b9b4eec9 Update pion/transport to v3.1.1 (#4070) b23d093c update protocol (#4069) 4ce07bed Higher resolution forwarding latency histogram. (#4067) 858db7ab fix(deps): update module github.com/livekit/protocol to v1.43.0 (#4015) 1dc9b8fc Use buffered indicator to exclude from forwarding latency. (#4062) f117ee51 Track start up delay. (#4061) 4872f205 Return write count from WriteRTP. (#4059) d0ba46b4 Log write count atomic. (#4057) ae5fb7e8 Add packet to forwarding stats only if packet is forwarded. (#4056) f6909192 Update PsRPC to get redis pipeliner implementation. (#4055) ca3c507b Prevent invalid track access while peer connection is shutting down. (#4054) 9ca6ee00 Use replace so that x/tools does not get overridden (#4048) b9323eab chore(deps): downgrade x/tools for counterfeiter (#4047) 2f1e6c36 Prep release v1.9.3 (#4046) 9d5c351d Fix prom units for forwarding latency/jitter. (#4045) e183657c Add prom histogram for forwarding latency and jitter. (#4044) 1eefeb30 Enable AbsCaptureTimeURI in RTC configuration (#4043) 075a7576 Use simulcast codec as default policy for audio track (#4040) c264b504 Don't warn 0 payload type for PCMU (#4039) 32fc3525 Broadcast cond var on RTX write. (#4038) 061eb8b4 AddDownTrack to regressed codec after restarting forwarder. (#4037) c87eb8ed fix: add missing Unlock() in AddReceiver (#4036) 70444924 if RingingTimeout is provided, deadline should be set to that timeout. (#4018) ab906d71 Prevent leakage of previous codec after codec regression. (#4035) 79b03f97 Log queueing latency when encountering high forwarding latency (#4034) 29117b14 set max layer in allocation (#4033) 15b19ccd Remove ~ from rid which indicates disabled layer to get the actual rid (#4032) 34e16a87 Check more conditions for opportunistic alloc. (#4031) 81fbd355 Use the optimal allocation function for opportunistic allocation. (#4030) a2ce73e0 Do not bind buffer if codec is invalid. (#4028) cef6fdb7 Correct direction for request/response for prom counters. (#4027) 5042c06c Use rtp converter from protocol/utils/rtputil (#4020) 5a426d15 Use rtp converter from protocol/utils (#4019) 35fb8877 feat: use env var for GOARCH (#4012) c0397696 Issue #1 only: Fix spatial layer initialization in Forwarder (#4003) 2afbf0e8 Some golang modernisation bits. (#4016) 484f784a Prepare release v1.9.2 (#4011) ad074ed2 counterfeiter needs an older version of x/tools (#4009) e63e8b6f Include mid -> trackID in both SDP offer and answer. (#4007) 781dfede Do not call receiver methods under settings lock. (#4006) 69ff25a0 Use answer with mid -> trackID mapping when in single peer connection (#4005) fe912acf Update pion/webrtc to prevent GetStats panic. (#4004) 7930dcde Do not try to read stats from peer connection after close. (#4002) ca0d5ee9 Count request/response packets on both client and server side. (#4001) dd62eb00 Resort to full search for requested quality is not available. (#4000) f6ca82d1 Revert to using silence packets for audio dummy start. (#3999) 0e2c59c8 Sort codec layers when adding track (#3998) 100bb46a Adding ProviderInfo to GetSIPTrunkAuthenticationResponse (#3993) a8d4df66 "Power of Two Random Choices" option for node selection (#3785) a20bbe34 Log RPC details. (#3991) 158496bc Increment RTP timestamp on padding when using dummy start. (#3989) 4f6ed65d Limit check to red + opus when looking for primary codec match. (#3988) a87f6c4b Allow passing inline trunk for outbound calls. (#3987) bf06596f Support Opus mixed with RED when encrypted. (#3986) ea208a1c Add encryption datapacket type (#3869) 2a6adbe8 Use padding only packets for dummy start of audio. (#3984) be018f97 Provide the InputVideo/AudioState to Ingress in WHIPRTCConnectionNotify (#3982) 146bd969 Do not panic of redis is not configured (#3981) 01337ba7 Do not start forawarding on out-of-order packet. (#3985) c7f625d6 Do not force codec regression between opus and red. (#3980) 3bd20ddb Revert unintentional change to not handle transport fallback on (#3970) 89a2f46c Update deps to fix redis issue when 1 cluster address is provided (#3969) 060719d1 add config for user data recording (#3966) b3ee219c fix stats worker closed condition (#3965) 3d737031 add idempotent reference count to telemetry stats worker (#3964) 735c663a Update protocol for EventKey helper. (#3963) 646b9de8 Add node_ip to config-sample.yaml (#3960) 0bf7b178 avoid logging on small values (#3958) 00ff2ab9 Adjust for hold time when fowarding RTCP report. (#3956) e180be06 short circuit participant broadcast filter in livestream mode (#3955) bfba6fee Adjust stream allocator ping interval based on state. (#3951) 3837006b Revert "Switch ops queue a singly linked list. (#3949)" (#3950) 990c5faf feat: server rpc apis (#3904) 80b11662 Switch ops queue a singly linked list. (#3949) 56ee2328 handle terminated job requests (#3948) 49f9b9c8 Flush stats when there are no packets. (#3947) e6a3df1e ForwarStats.GetStats needs to be public (#3946) 824d116b Tweaks tresholds for logging high forwarding latency/jitter. (#3945) 408492e0 Log some information around high forwarding latency. (#3944) 6a41fae5 Use microseconds for forwarding stats. (#3943) 856e0871 mediatransportutil to log local address when validating external IP (#3942) 40101cf7 Update protocol for SipCreateParticipant (#3939) b07e7a38 Use difference in key frame counter to stop seeder. (#3936) d7f92878 Avoid matching on empty track id. (#3937) 56fb2885 Do DD restart only if DD structure is present. (#3935) 86facce9 More debugging of DD jump (#3934) 6058a3f6 Add debugging from DD frame number wrap around. (#3933) dc6825c0 mediatransportutil crash fix for logging local address (#3930) d6f0588f Update mediatransportutil to log external IP found via STUN. (#3929) 2c30a064 Fix dynacast subscriber node clearing on move participant. (#3926) 6489237e Simulcast audio fixes (#3925) 9f0ab870 Wait for `SetRemoteDescription` before configuring senders. (#3924) df6c26db Subscrbed audio codecs - update from remote nodes. (#3921) 798fa761 Support simulcasting of audio (#3920) f4a06cf0 Clean code as there is no oss sweeper for ingress (#3918) 5f561b4f Include agent_name as a participant attribute (#3914) 782a35e8 update protocol for psrpc (#3915) eee2001a Set publisher codec preferences after setting remote description (#3913) fc995533 add incoming request id to request response message (#3912) 76645fad Rpcs for ingress proxy WHIP (#3911) 991a4a4f Refactor subscribedTrack + mediaTrackSubscriptions. (#3908) e16b3ba9 Use gzip reader pool (#3903) 17c34921 update protocol for sip api change (#3902) 2f43a575 Release candidate for v1.9.1 (#3899) 07c40cf3 Use `RequestResponse` to report protocol handling errors (#3895) 98352fd0 Prevent race in determining BWE type. (#3891) f7291fda Do not send both asb-send-time and twcc. (#3890) 21b42fa6 Do not advertise NACK for RED. (#3889) 6633bf93 Use departure timeout from room preset. (#3888) 38f7906e Handle migration better in single peer connection case. (#3886) 5026de2b handle frame number wrap back in svc (#3885) 091e3c13 Revert to using answer for migration case. (#3884) 2aeadf14 init ua parser once (#3883) 998a9f94 Switch known rids from 012 -> 210, used by OBS. (#3882) 890fd942 Single peer connection mode (#3873) bfe98eaa fix: ensure the participant kind is set on refresh tokens (#3881) 8d270e2a chunk room updates (#3880) b4e146c5 update mediatransport util for ice port 3478 (#3877) dc3a7753 Fix timeout handing in StopEgress (#3876) d62336e1 Remove unnecessary check (#3870) c58e5d23 Update golang Docker tag to v1.25 (#3864) 98d577ee Update module github.com/livekit/protocol to v1.40.0 (#3865) afbf541e Update pion deps (#3863) b660c3b5 Extract video size from media stream (#3856) 456b8709 Fix missed unlock (#3861) d500806e Handle no codecs in track info. (#3859) 11b240d6 Log track settings more. (#3853) 1aa0f963 Log signal messages on media node. (#3852) b182d07b Log signal messages as debug. (#3851) a370bb20 Support G.711 A-law and U-law (#3849) fa5f4ef3 Populate SDP cid in track info when available. (#3845) eed27885 Send `participant_connection_aborted` when participant session is closed (#3848) 61e59346 Update go deps (#3439) 1b228913 Support video layer mode from client and make most of the code mime aware (#3843) f2da4444 Support per simulcast codec layers. (#3840) f275f592 handle SyncState in join request (#3839) 5d44cf6d Use wrapped join request to be able to support compressed and uncompressed. (#3838) 5ca16264 Support join request as proto + base64 encoded query param (#3836) 7dea1012 Clean up missed v2 pieces (#3837) 34a49130 Delete v2 signalling (#3835) 1fe33716 Fix: RingingTimeout was being skipped for transferParticipant (#3831) 5751692a deps (#3829) db4bc127 Get to the point of connecting publisher PC and using it for async signalling (#3822) 5e483e75 update readme (#3809) e3155b14 Get to the point of establishing subscriber peer connection. (#3821) a7ce1382 HTTP DELETE of participant session (#3819) 01de0e36 Do not send leave if nil (to older clients) (#3817) 10103449 Add country label to edge prom stats. (#3816) 68387b41 Minor tweak to keep RPC type at service level. (#3815) a75295fc More v2 signalling changes (#3814) b20db94d Validation end point for v2 signalling. (#3811) f2f595f4 update readme (#3808) fffc2ac0 Use signalling utils from protocol (#3807) f5fc82d3 Filling out messages unlikely to change in v2. (#3806) 1c99b9ad Split signal segmenter and reassembler. (#3805) 0a1bfd30 Signal handling interfaces and participant specific HTTP PATCH. (#3804) 7837c8e5 starting signaller interface (#3802) 18ce5244 Handle Metadata field from RoomConfig (#3798) 2a6a9b8a Grouping all signal messages into participant_signal. (#3801) 078c01fa Signal v2: envelope and fragments as wire message format. (#3800) b9a44c3f Signalling V2 protocol implementation start (#3794) ba702a53 forward agent id to job state (#3786) 1f31d430 Map ErrNoResponse to ErrRequestTimedOut in StopEgress to avoid returning 503 (#3788) 51bbe8c5 Set participant active when peerconnection connected (#3790) 40028dc3 Normalize known rids. (#3779) ddd92329 Return default layer for invalid rid + track info combination. (#3778) 8c033ce9 Enable H265 by default (#3773) 7678e087 Set rids for all codecs. (#3772) 5d636acf Limit taking rids from SDP only in WHIP path. (#3771) 4d09e5b5 Log SDP rids to understand the mapping better. (#3770) c69f1aae Revert "Temporary change: use pre-defined rids" (#3769) 8197438e bounds check layer index (#3768) d11da5f5 Temporary change: use pre-defined rids (#3767) cb4da533 fix(deps): update module github.com/livekit/protocol to v1.39.3 (#3733) d6d2b6d8 feat(cli-flags): add option for cpu profiling (#3765) 9fc4ddbe ClearAllReceivers interface is used to pause relay tracks. (#3761) 1216113b Do not need to just clean up receivers. Remove that interface. (#3760) ef6c38ce Log previous allocation to see changes. (#3759) 01bf9685 SVC with RID -> spatial layer mapping (#3754) c481396f offer could be nil when migrating. (#3752) 8c2fc0bc Fix svc encoding for chrome mobile on iOS (#3751) e467daa0 move egress roomID load to launcher (#3748) 3783ebb3 feat(cli): update to urfave/cli/v3 (#3745) 03d3fcab Fix data packet ParticipantIdentity override logic in participant.go (#3735) 068b4366 reuse compiled client config scripts (#3743) e754a860 return error when moving egree/agent participant (#3741) 7542cf07 remove unused code (#3740) 9d569e2f Take ClientInfo from request. (#3738) 80774263 chore: set workerid on job creation (#3737) 5549ab55 Revert clearing RIDs. (#3732) ae967313 Clear rids if not present in SDP. (#3731) 0e033907 Return highest available layer if requested quality is higher than (#3729) 9ce737db Add log for dropping out of order reliable message (#3728) 1b95e818 Don't check bindState on downtrack.Bind (#3726) 670f927f Set and use rid/spatial layer in TrackInfo. (#3724) a9e29116 Add Id to SDP signalling messages. (#3722) 4ec828ce Fix bug with SDP rid, clear only overflow. (#3723) 8f6c3a9b Clear rids from default for layers not published. (#3721) ce07740e Add simulcast support for WHIP. (#3719) e98fb94f Create client config manager in room manager constructor. (#3718) fdf9b852 e2e reliability for data channel (#3716) 35dda8ea swap pub/sub track metrics (#3717) 1d9a4366 Do not require create permission for WHIP participant. (#3715) e0aea17a Flush stats on close (#3713) 630aa7d9 implement observability for room metrics (#3712) e7f0294e remove unused ws signal read loop (#3709) 77f70b18 for real, pick up protocol change for webhooks queue length bucnkets (#3707) 7b180646 protocol dep for webhook stats buckets (#3706) b0ab95ba warn about credentials when used in tokens (#3705) a72ce30f Small changes to add/use helper functions for length checks. (#3704) 425f6bb3 Allow passing extra attributes to RTC endpoint. (#3693) 758e1762 Add a trend check before declaring joint queuing region. (#3701) fe81e411 Adds Devin to readme so it auto updates DeepWiki weekly (#3699) 09dede35 version bump v1.9.0 (#3698) fc867c5b Webhook prom stats (#3697) 0e17916f Do not use Redis pipeline for SIP delete. Fixes Redis clustering support. (#3694) 1b760393 WHIP support. (#3692) e4f7d81b add client ip to agent worker registration (#3675) 6b849a87 update mediatransportutil for sctp congestion control (#3673) 83b189b0 Add ServerInfo to ReconnectResponse (#3671) 5f87a35b Prevent operating on swapped out map. (#3670) 13b55a80 move agent token (#3669) c9385edd handle agent worker jwt (#3668) 3b359d8b Use logger resolver reset to reset contexts. (#3665) dbb70e0f Fix dynacast quality for moving out tracks (#3664) 0a5f3c2a resolve new room name logger earlier when moving participant (#3662) 2df05517 Revert unbound transceiver stop. (#3661) 5172af15 ~Send initial participant update only after a participant becomes active.~ - General clean up (#3655) 5d4d86f8 protocol to pick resolver values replace (#3659) 7f8e6323 Send self participant update immediately. (#3656) 11630878 Use unordered for lossy data channel. (#3653) aee34ffe log request for agent dispatch api (#3650) 793b383a Add Moving participant to another room (#3648) 2fff36cb Stub MoveParticipant so that cloud can include the latest protocol. (#3646) d4ab1142 Redact address (#3643) b83190a3 protocol update to fix memory stats path (#3642) a1f4e88e Update protocol to latest, got bit by tag (#3641) 8d3902af Protocol to pick up cgroups v2 memory path fix (#3640) 2f002388 Use participant close reason in remove. (#3639) 58822c26 Include clientInfo in connectivity logs. (#3638) 6d6393a6 Take AudioFeatures from AddTrack. (#3635) 08670412 Limit buffer queue before Bind. (#3634) 9f5bc9b9 Avoid synthesising duplicate feature. (#3632) 847239c3 Disable vp9 for safari 18.4 (#3631) f69ab680 Populate the sender identity when translating to user packet. (#3628) e1490558 Forward data between WHIP client and non-WHIP client (#3627) 6739e7bc Broadcast inside lock (#3626) f24152b4 Call Broadcast in lock scope. (#3625) b760918a Use logger from request context. (#3623) 34a2e2c1 Check for multiple layers for managed track. (#3622) 4955ebe4 Forward transfer headers to internal request (#3615) d9ee9214 Set up RTX for WHIP publish (#3619) d8cf5439 Determine TURN connection type and no fallback for TURN/TLS. (#3612) d0d212fd Fix WHIP ICE restart. (#3616) 5e7f8a12 Update mediatransportutil for max sctp message (65K) (#3611) 28dfac14 Use exported GetEgressNotifyOptions (#3604) 75236bef protocol update to fix IPv6 SDP fragment parsing (#3603) 2130980d Add basic video support to WHIP. (#3602) e5cbb227 Allow specifying extra webhooks with egress requests (#3597) 7e16106a Add OnSubscirberReady callback on LocalParticipant. (#3600) 5c2d96b9 Check DestinationRoom of VideoGrant for participant forwarding (#3599) 2e236a19 Revert participant state ACTIVE change. (#3598) 35ac5f56 Add support for WHIP ICE Trickle/Restart. (#3596) ec2dff96 Fix SIP updates when replacing slices. (#3592) e24fe77b map PEER_CONNECTION_DISCONNECTED -> CONNECTION_TIMEOUT (#3591) 6ee6eb43 Do not drop audio codecs (#3590) 68357ba6 List audio codecs after video codecs. (#3589) 05a891ff Fix rule (had an extra bracket) (#3588) d7c41091 Exclude RED from enabled codecs for Flutter + 2.4.2 + Android. (#3587) ee08aede skip out of order participant state updates (#3583) 15a8d9a2 Break track published fuse when there are no tracks (#3581) 35f83c51 Replace Promise with Fuse. (#3580) 07fe9b72 Prevent migration race. (#3579) ac8082ff Use older SDP module to accommodate bad SDP. (#3578) 1c8307c7 Use cgroup for memstats. (#3573) e9be0fca log SDP offer on error (#3577) 3238ab8d Calculate rates for memory used and total. (#3570) d08487bf Unlabeled (pass through) data channels. (#3567) 52ce18d5 fix: revert recent changes to determine simulcast from sdp (#3565) cdfbb106 Audio uses signal SignalCid and SdpCid. (#3564) ed5e2f16 Keep simulcast information tied to receiver. (#3563) ad010cfc chore(logs): log VLS type for VP9/AV1 (#3561) 8cc17f8f Rework node stats a bit. (#3555) 15f56551 fix(video): determine svc/simulcast from SDP for advanced codecs (#3549) 2b6a46f4 Handle `prefer_regression` for backup codec (#3554) b0abb0ae Add option to use different pacer with send side bwe. (#3552) 26822b6b ParseUsername utility for TURN user name. (#3547) 55909ed7 log the initial join response (#3546) 97fcb82a Fix: Return NotFoundErr instead of Unavailable when the participant does not exist in UpdateParticipant. (#3543) 75d0e18e Implement SIP update API. (#3141) e118aff1 Fire track subscribed when the subscriber connected (#3540) 13417c01 Send mute event only on change (#3537) 7f4c4597 Stubs for SIP update API. (#3533) fe673bb2 Send regressed codec upstream stats to analytics. (#3532) 8eb81388 Use a generation to counter to stop key frame seeder on codec change (#3531) 188470a2 Do not accept unsupported track type in AddTrack (#3530) 507fc9cf Do not instantiate 0 sized sequencer. (#3529) 20bddfea Clean up published track on participant removal. (#3527) 65d8aa28 Handle subscribe race with track close better. (#3526) a6cb00b3 Reduce seeder duration to 30s and also do not force send PLI. (#3525) c8233205 Add a key frame seeder in up track. (#3524) 0f61ff3a Remove redundant log (#3523) 7685cd25 Log ParticipantInit on signal start to get a picture of join params (#3522) ac9e62ef add server agent load threshold config (#3520) cd5d32f0 Add pID and connID to log context to make it easier to search using pID. (#3518) 2d9aa6dd Update api call info method (#3515) b3779a90 WebHookConfig (#3517) 6121b9af Check ForwardParticipant room name (#3514) 9a7c9442 mediatransportutil update (#3511) 50ab47c1 Log packet drops/forward. (#3510) 139d1b13 Add ForwardParticipant method to room service (#3507) 6c04909f Use atomic to store codec. (#3505) 7f6afe05 Prevent bind lock deadlock on muted. (#3504) 48063df5 load mime type before calling writeBlankFrameRTP (#3502) d2e6cd15 Do not bind lock across flush which could take time (#3501) 47896f50 Update protocol and IO service. (#3499) 1dc42eef Bump github.com/go-jose/go-jose/v3 from 3.0.3 to 3.0.4 (#3497) 3a35cbc4 Log migration complete only when coming from sync (#3496) c2f17a10 refactor: using slices.Contains to simplify the code (#3495) 01e51dbd fix: fix the wrong error return value (#3493) ff9115b2 Disable dd parser for vp8 if extension is not found (#3492) c3e06f05 Do not attempt to create objects for URL ingresses as the ingress service will do so (#3491) f0edfbba Fix receiver rtt/jitter. (#3487) 05dfd30d Take RTT and jitter from receiver view while reporting track stats for (#3483) 04ed5683 Don't issue TrackPublished/Unpublished event on migrated track (#3482) 1cffe30c Use a RED transformer to consolidate both RED -> Opus OR Opus -> RED (#3481) 591888f7 Fix missing RTCP sender report when forwarding RED as Opus. (#3480) git-subtree-dir: livekit-server git-subtree-split: 46c4309554d37d23ee8da88a8a7e02a68fba09c1
1515 lines
45 KiB
Go
1515 lines
45 KiB
Go
// Copyright 2023 LiveKit, Inc.
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//
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// Licensed under the Apache License, Version 2.0 (the "License");
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// you may not use this file except in compliance with the License.
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// You may obtain a copy of the License at
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//
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// http://www.apache.org/licenses/LICENSE-2.0
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//
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// Unless required by applicable law or agreed to in writing, software
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// distributed under the License is distributed on an "AS IS" BASIS,
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// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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// See the License for the specific language governing permissions and
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// limitations under the License.
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package test
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import (
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"context"
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"encoding/binary"
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"errors"
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"fmt"
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"net"
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"net/http"
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"reflect"
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"strings"
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"sync"
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"testing"
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"time"
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"github.com/jxskiss/base62"
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"github.com/pion/sdp/v3"
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"github.com/pion/stun/v3"
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"github.com/pion/turn/v5"
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"github.com/pion/webrtc/v4"
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"github.com/stretchr/testify/require"
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"github.com/thoas/go-funk"
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"github.com/twitchtv/twirp"
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"go.uber.org/atomic"
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"github.com/livekit/protocol/auth"
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"github.com/livekit/protocol/codecs/mime"
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"github.com/livekit/protocol/livekit"
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"github.com/livekit/protocol/logger"
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"github.com/livekit/livekit-server/pkg/config"
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"github.com/livekit/livekit-server/pkg/rtc"
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"github.com/livekit/livekit-server/pkg/rtc/types"
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"github.com/livekit/livekit-server/pkg/service"
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"github.com/livekit/livekit-server/pkg/sfu"
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"github.com/livekit/livekit-server/pkg/sfu/datachannel"
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"github.com/livekit/livekit-server/pkg/testutils"
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testclient "github.com/livekit/livekit-server/test/client"
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)
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const (
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waitTick = 10 * time.Millisecond
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waitTimeout = 5 * time.Second
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)
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func TestClientCouldConnect(t *testing.T) {
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if testing.Short() {
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t.SkipNow()
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return
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}
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_, finish := setupSingleNodeTest("TestClientCouldConnect")
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defer finish()
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for _, testRTCServicePath := range testRTCServicePaths {
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t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
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c1 := createRTCClient("c1", defaultServerPort, testRTCServicePath, nil)
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c2 := createRTCClient("c2", defaultServerPort, testRTCServicePath, nil)
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waitUntilConnected(t, c1, c2)
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// ensure they both see each other
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testutils.WithTimeout(t, func() string {
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if len(c1.RemoteParticipants()) == 0 {
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return "c1 did not see c2"
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}
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if len(c2.RemoteParticipants()) == 0 {
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return "c2 did not see c1"
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}
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return ""
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})
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})
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}
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}
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func TestClientConnectDuplicate(t *testing.T) {
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if testing.Short() {
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t.SkipNow()
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return
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}
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_, finish := setupSingleNodeTest("TestClientConnectDuplicate")
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defer finish()
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for _, testRTCServicePath := range testRTCServicePaths {
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t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
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grant := &auth.VideoGrant{RoomJoin: true, Room: testRoom}
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grant.SetCanPublish(true)
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grant.SetCanSubscribe(true)
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token := joinTokenWithGrant("c1", grant)
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c1 := createRTCClientWithToken(token, defaultServerPort, testRTCServicePath, nil)
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// publish 2 tracks
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t1, err := c1.AddStaticTrack("audio/opus", "audio", "webcam")
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require.NoError(t, err)
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defer t1.Stop()
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t2, err := c1.AddStaticTrack("video/vp8", "video", "webcam")
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require.NoError(t, err)
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defer t2.Stop()
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c2 := createRTCClient("c2", defaultServerPort, testRTCServicePath, nil)
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waitUntilConnected(t, c1, c2)
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opts := &testclient.Options{
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Publish: "duplicate_connection",
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}
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testutils.WithTimeout(t, func() string {
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if len(c2.SubscribedTracks()) == 0 {
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return "c2 didn't subscribe to anything"
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}
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// should have received two tracks
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if len(c2.SubscribedTracks()[c1.ID()]) != 2 {
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return "c2 didn't subscribe to both tracks from c1"
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}
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// participant ID can be appended with '#..' . but should contain orig id as prefix
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tr1 := c2.SubscribedTracks()[c1.ID()][0]
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participantId1, _ := rtc.UnpackStreamID(tr1.StreamID())
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require.Equal(t, c1.ID(), participantId1)
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tr2 := c2.SubscribedTracks()[c1.ID()][1]
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participantId2, _ := rtc.UnpackStreamID(tr2.StreamID())
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require.Equal(t, c1.ID(), participantId2)
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return ""
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})
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c1Dup := createRTCClientWithToken(token, defaultServerPort, testRTCServicePath, opts)
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waitUntilConnected(t, c1Dup)
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t3, err := c1Dup.AddStaticTrack("video/vp8", "video", "webcam")
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require.NoError(t, err)
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defer t3.Stop()
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testutils.WithTimeout(t, func() string {
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if len(c2.SubscribedTracks()[c1Dup.ID()]) != 1 {
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return "c2 was not subscribed to track from duplicated c1"
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}
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tr3 := c2.SubscribedTracks()[c1Dup.ID()][0]
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participantId3, _ := rtc.UnpackStreamID(tr3.StreamID())
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require.Contains(t, c1Dup.ID(), participantId3)
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return ""
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})
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})
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}
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}
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func TestSinglePublisher(t *testing.T) {
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if testing.Short() {
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t.SkipNow()
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return
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}
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s, finish := setupSingleNodeTest("TestSinglePublisher")
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defer finish()
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for _, testRTCServicePath := range testRTCServicePaths {
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t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
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c1 := createRTCClient("c1", defaultServerPort, testRTCServicePath, nil)
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c2 := createRTCClient("c2", defaultServerPort, testRTCServicePath, nil)
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waitUntilConnected(t, c1, c2)
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// publish an audio and video track and ensure clients receive it ok
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t1, err := c1.AddStaticTrack("audio/OPUS", "audio", "webcamaudio")
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require.NoError(t, err)
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defer t1.Stop()
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t2, err := c1.AddStaticTrack("video/vp8", "video", "webcamvideo")
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require.NoError(t, err)
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defer t2.Stop()
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testutils.WithTimeout(t, func() string {
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if len(c2.SubscribedTracks()) == 0 {
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return "c2 was not subscribed to anything"
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}
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// should have received two tracks
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if len(c2.SubscribedTracks()[c1.ID()]) != 2 {
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return "c2 didn't subscribe to both tracks from c1"
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}
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tr1 := c2.SubscribedTracks()[c1.ID()][0]
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participantId, _ := rtc.UnpackStreamID(tr1.StreamID())
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require.Equal(t, c1.ID(), participantId)
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return ""
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})
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// ensure mime type is received
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remoteC1 := c2.GetRemoteParticipant(c1.ID())
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audioTrack := funk.Find(remoteC1.Tracks, func(ti *livekit.TrackInfo) bool {
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return ti.Name == "webcamaudio"
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}).(*livekit.TrackInfo)
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require.Equal(t, "audio/opus", audioTrack.MimeType)
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// a new client joins and should get the initial stream
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c3 := createRTCClient("c3", defaultServerPort, testRTCServicePath, nil)
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// ensure that new client that has joined also received tracks
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waitUntilConnected(t, c3)
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testutils.WithTimeout(t, func() string {
|
|
if len(c3.SubscribedTracks()) == 0 {
|
|
return "c3 didn't subscribe to anything"
|
|
}
|
|
// should have received two tracks
|
|
if len(c3.SubscribedTracks()[c1.ID()]) != 2 {
|
|
return "c3 didn't subscribe to tracks from c1"
|
|
}
|
|
return ""
|
|
})
|
|
|
|
// ensure that the track ids are generated by server
|
|
tracks := c3.SubscribedTracks()[c1.ID()]
|
|
for _, tr := range tracks {
|
|
require.True(t, strings.HasPrefix(tr.ID(), "TR_"), "track should begin with TR")
|
|
}
|
|
|
|
// when c3 disconnects, ensure subscriber is cleaned up correctly
|
|
c3.Stop()
|
|
|
|
testutils.WithTimeout(t, func() string {
|
|
room := s.RoomManager().GetRoom(context.Background(), testRoom)
|
|
p := room.GetParticipant("c1")
|
|
require.NotNil(t, p)
|
|
|
|
for _, t := range p.GetPublishedTracks() {
|
|
if t.IsSubscriber(c3.ID()) {
|
|
return "c3 was not a subscriber of c1's tracks"
|
|
}
|
|
}
|
|
return ""
|
|
})
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestConnectionStats(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
s, finish := setupSingleNodeTest("TestConnectionStats")
|
|
defer finish()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
c1 := createRTCClient("c1", defaultServerPort, testRTCServicePath, nil)
|
|
c2 := createRTCClient("c2", defaultServerPort, testRTCServicePath, nil)
|
|
waitUntilConnected(t, c1, c2)
|
|
defer func() {
|
|
c1.Stop()
|
|
c2.Stop()
|
|
}()
|
|
|
|
// both clients publish audio + video
|
|
t1, err := c1.AddStaticTrack("audio/opus", "audio", "c1audio")
|
|
require.NoError(t, err)
|
|
defer t1.Stop()
|
|
t2, err := c1.AddStaticTrack("video/vp8", "video", "c1video")
|
|
require.NoError(t, err)
|
|
defer t2.Stop()
|
|
|
|
t3, err := c2.AddStaticTrack("audio/opus", "audio", "c2audio")
|
|
require.NoError(t, err)
|
|
defer t3.Stop()
|
|
t4, err := c2.AddStaticTrack("video/vp8", "video", "c2video")
|
|
require.NoError(t, err)
|
|
defer t4.Stop()
|
|
|
|
// wait for cross-subscriptions: each client should receive 2 tracks from the other
|
|
testutils.WithTimeout(t, func() string {
|
|
if len(c1.SubscribedTracks()[c2.ID()]) != 2 {
|
|
return "c1 did not subscribe to both tracks from c2"
|
|
}
|
|
if len(c2.SubscribedTracks()[c1.ID()]) != 2 {
|
|
return "c2 did not subscribe to both tracks from c1"
|
|
}
|
|
return ""
|
|
})
|
|
|
|
room := s.RoomManager().GetRoom(context.Background(), testRoom)
|
|
require.NotNil(t, room)
|
|
|
|
// hook the upstream WebRTCReceiver.OnStatsUpdate and downstream DownTrack.OnStatsUpdate
|
|
// callbacks so we can verify the AnalyticsStat delivered through each carries valid
|
|
// delta data. MediaTrack.Receivers() returns one entry per potential codec; only those
|
|
// matching the actually published codec are *sfu.WebRTCReceiver, the rest are
|
|
// placeholder *rtc.DummyReceiver instances that we skip.
|
|
type statCapture struct {
|
|
lock sync.Mutex
|
|
stat *livekit.AnalyticsStat
|
|
}
|
|
receiverCaptures := make(map[livekit.TrackID]*statCapture)
|
|
downTrackCaptures := make(map[livekit.ParticipantIdentity]map[livekit.TrackID]*statCapture)
|
|
for _, identity := range []livekit.ParticipantIdentity{"c1", "c2"} {
|
|
p := room.GetParticipant(identity)
|
|
require.NotNil(t, p, "participant %s not found", identity)
|
|
for _, mt := range p.GetPublishedTracks() {
|
|
rc := &statCapture{}
|
|
receiverCaptures[mt.ID()] = rc
|
|
var hooked int
|
|
for _, r := range mt.Receivers() {
|
|
if dr, ok := r.(*rtc.DummyReceiver); ok {
|
|
underlying := dr.Receiver()
|
|
if underlying == nil {
|
|
continue
|
|
}
|
|
r = underlying
|
|
}
|
|
wr, ok := r.(*sfu.WebRTCReceiver)
|
|
if !ok {
|
|
continue
|
|
}
|
|
wr.OnStatsUpdate(func(_ *sfu.WebRTCReceiver, stat *livekit.AnalyticsStat) {
|
|
rc.lock.Lock()
|
|
rc.stat = stat
|
|
rc.lock.Unlock()
|
|
})
|
|
hooked++
|
|
}
|
|
require.Greater(t, hooked, 0, "no live WebRTCReceiver found for published track %s", mt.ID())
|
|
}
|
|
|
|
dtCaps := make(map[livekit.TrackID]*statCapture)
|
|
downTrackCaptures[identity] = dtCaps
|
|
for _, st := range p.GetSubscribedTracks() {
|
|
dt := st.DownTrack()
|
|
require.NotNil(t, dt, "subscribed track %s has no DownTrack", st.ID())
|
|
dc := &statCapture{}
|
|
dtCaps[st.ID()] = dc
|
|
dt.OnStatsUpdate(func(_ *sfu.DownTrack, stat *livekit.AnalyticsStat) {
|
|
dc.lock.Lock()
|
|
dc.stat = stat
|
|
dc.lock.Unlock()
|
|
})
|
|
}
|
|
}
|
|
|
|
validateAnalyticsStat := func(stat *livekit.AnalyticsStat) string {
|
|
if stat == nil {
|
|
return "stat nil"
|
|
}
|
|
if len(stat.Streams) == 0 {
|
|
return "stat has no streams"
|
|
}
|
|
var totalPackets uint32
|
|
var totalBytes uint64
|
|
for _, s := range stat.Streams {
|
|
totalPackets += s.PrimaryPackets
|
|
totalBytes += s.PrimaryBytes
|
|
}
|
|
if totalPackets == 0 {
|
|
return "stat has no packets across streams"
|
|
}
|
|
if totalBytes == 0 {
|
|
return "stat has no bytes across streams"
|
|
}
|
|
return ""
|
|
}
|
|
|
|
// wait for cumulative + delta + OnStatsUpdate-derived stats. the
|
|
// connection-quality update interval is 5s, so allow plenty of time for
|
|
// the receiver OnStatsUpdate callback to fire at least once and for
|
|
// the downstream connection-quality scorer to compute a real score.
|
|
testutils.WithTimeout(t, func() string {
|
|
for _, identity := range []livekit.ParticipantIdentity{"c1", "c2"} {
|
|
p := room.GetParticipant(identity)
|
|
if p == nil {
|
|
return fmt.Sprintf("participant %s not found", identity)
|
|
}
|
|
|
|
// upstream (publisher) cumulative stats
|
|
published := p.GetPublishedTracks()
|
|
if len(published) != 2 {
|
|
return fmt.Sprintf("%s expected 2 published tracks, got %d", identity, len(published))
|
|
}
|
|
for _, mt := range published {
|
|
lmt, ok := mt.(types.LocalMediaTrack)
|
|
if !ok {
|
|
return fmt.Sprintf("%s published track %s is not a LocalMediaTrack", identity, mt.ID())
|
|
}
|
|
stats := lmt.GetTrackStats()
|
|
if stats == nil {
|
|
return fmt.Sprintf("%s upstream cumulative stats nil for track %s", identity, mt.ID())
|
|
}
|
|
if stats.Packets == 0 {
|
|
return fmt.Sprintf("%s upstream cumulative stats has no packets for track %s", identity, mt.ID())
|
|
}
|
|
if stats.Bytes == 0 {
|
|
return fmt.Sprintf("%s upstream cumulative stats has no bytes for track %s", identity, mt.ID())
|
|
}
|
|
|
|
// upstream delta stats fed into the receiver OnStatsUpdate path
|
|
rc, ok := receiverCaptures[mt.ID()]
|
|
if !ok {
|
|
return fmt.Sprintf("%s missing receiver capture for track %s", identity, mt.ID())
|
|
}
|
|
rc.lock.Lock()
|
|
stat := rc.stat
|
|
rc.lock.Unlock()
|
|
if msg := validateAnalyticsStat(stat); msg != "" {
|
|
return fmt.Sprintf("%s upstream OnStatsUpdate %s for track %s", identity, msg, mt.ID())
|
|
}
|
|
}
|
|
|
|
// downstream (subscriber) cumulative stats and DownTrack OnStatsUpdate
|
|
// delta stats captured from the listener path
|
|
subscribed := p.GetSubscribedTracks()
|
|
if len(subscribed) != 2 {
|
|
return fmt.Sprintf("%s expected 2 subscribed tracks, got %d", identity, len(subscribed))
|
|
}
|
|
for _, st := range subscribed {
|
|
dt := st.DownTrack()
|
|
if dt == nil {
|
|
return fmt.Sprintf("%s subscribed track %s has no DownTrack", identity, st.ID())
|
|
}
|
|
stats := dt.GetTrackStats()
|
|
if stats == nil {
|
|
return fmt.Sprintf("%s downstream cumulative stats nil for track %s", identity, st.ID())
|
|
}
|
|
if stats.Packets == 0 {
|
|
return fmt.Sprintf("%s downstream cumulative stats has no packets for track %s", identity, st.ID())
|
|
}
|
|
if stats.Bytes == 0 {
|
|
return fmt.Sprintf("%s downstream cumulative stats has no bytes for track %s", identity, st.ID())
|
|
}
|
|
|
|
// downstream delta stats fed into the DownTrack OnStatsUpdate path
|
|
dc, ok := downTrackCaptures[identity][st.ID()]
|
|
if !ok {
|
|
return fmt.Sprintf("%s missing DownTrack capture for track %s", identity, st.ID())
|
|
}
|
|
dc.lock.Lock()
|
|
stat := dc.stat
|
|
dc.lock.Unlock()
|
|
if msg := validateAnalyticsStat(stat); msg != "" {
|
|
return fmt.Sprintf("%s downstream OnStatsUpdate %s for track %s", identity, msg, st.ID())
|
|
}
|
|
}
|
|
}
|
|
return ""
|
|
}, 15*time.Second)
|
|
})
|
|
}
|
|
}
|
|
|
|
func Test_WhenAutoSubscriptionDisabled_ClientShouldNotReceiveAnyPublishedTracks(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
_, finish := setupSingleNodeTest("Test_WhenAutoSubscriptionDisabled_ClientShouldNotReceiveAnyPublishedTracks")
|
|
defer finish()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
opts := testclient.Options{AutoSubscribe: false}
|
|
publisher := createRTCClient("publisher", defaultServerPort, testRTCServicePath, &opts)
|
|
client := createRTCClient("client", defaultServerPort, testRTCServicePath, &opts)
|
|
defer publisher.Stop()
|
|
defer client.Stop()
|
|
waitUntilConnected(t, publisher, client)
|
|
|
|
track, err := publisher.AddStaticTrack("audio/opus", "audio", "webcam")
|
|
require.NoError(t, err)
|
|
defer track.Stop()
|
|
|
|
time.Sleep(syncDelay)
|
|
|
|
require.Empty(t, client.SubscribedTracks()[publisher.ID()])
|
|
})
|
|
}
|
|
}
|
|
|
|
func Test_RenegotiationWithDifferentCodecs(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
_, finish := setupSingleNodeTest("TestRenegotiationWithDifferentCodecs")
|
|
defer finish()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
c1 := createRTCClient("c1", defaultServerPort, testRTCServicePath, nil)
|
|
c2 := createRTCClient("c2", defaultServerPort, testRTCServicePath, nil)
|
|
waitUntilConnected(t, c1, c2)
|
|
|
|
// publish a vp8 video track and ensure clients receive it ok
|
|
t1, err := c1.AddStaticTrack("audio/opus", "audio", "webcam")
|
|
require.NoError(t, err)
|
|
defer t1.Stop()
|
|
t2, err := c1.AddStaticTrack("video/vp8", "video", "webcam")
|
|
require.NoError(t, err)
|
|
defer t2.Stop()
|
|
|
|
testutils.WithTimeout(t, func() string {
|
|
if len(c2.SubscribedTracks()) == 0 {
|
|
return "c2 was not subscribed to anything"
|
|
}
|
|
// should have received two tracks
|
|
if len(c2.SubscribedTracks()[c1.ID()]) != 2 {
|
|
return "c2 was not subscribed to tracks from c1"
|
|
}
|
|
|
|
tracks := c2.SubscribedTracks()[c1.ID()]
|
|
for _, t := range tracks {
|
|
if mime.IsMimeTypeStringVP8(t.Codec().MimeType) {
|
|
return ""
|
|
|
|
}
|
|
}
|
|
return "did not receive track with vp8"
|
|
})
|
|
|
|
t3, err := c1.AddStaticTrackWithCodec(webrtc.RTPCodecCapability{
|
|
MimeType: "video/h264",
|
|
ClockRate: 90000,
|
|
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
|
|
}, "videoscreen", "screen")
|
|
defer t3.Stop()
|
|
require.NoError(t, err)
|
|
|
|
testutils.WithTimeout(t, func() string {
|
|
if len(c2.SubscribedTracks()) == 0 {
|
|
return "c2's not subscribed to anything"
|
|
}
|
|
// should have received three tracks
|
|
if len(c2.SubscribedTracks()[c1.ID()]) != 3 {
|
|
return "c2's not subscribed to 3 tracks from c1"
|
|
}
|
|
|
|
var vp8Found, h264Found bool
|
|
tracks := c2.SubscribedTracks()[c1.ID()]
|
|
for _, t := range tracks {
|
|
if mime.IsMimeTypeStringVP8(t.Codec().MimeType) {
|
|
vp8Found = true
|
|
} else if mime.IsMimeTypeStringH264(t.Codec().MimeType) {
|
|
h264Found = true
|
|
}
|
|
}
|
|
if !vp8Found {
|
|
return "did not receive track with vp8"
|
|
}
|
|
if !h264Found {
|
|
return "did not receive track with h264"
|
|
}
|
|
return ""
|
|
})
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestSingleNodeRoomList(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
_, finish := setupSingleNodeTest("TestSingleNodeRoomList")
|
|
defer finish()
|
|
|
|
roomServiceListRoom(t)
|
|
}
|
|
|
|
func TestSingleNodeUpdateParticipant(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
_, finish := setupSingleNodeTest("TestSingleNodeRoomList")
|
|
defer finish()
|
|
|
|
adminCtx := contextWithToken(adminRoomToken(testRoom))
|
|
t.Run("update nonexistent participant", func(t *testing.T) {
|
|
_, err := roomClient.UpdateParticipant(adminCtx, &livekit.UpdateParticipantRequest{
|
|
Room: testRoom,
|
|
Identity: "nonexistent",
|
|
Permission: &livekit.ParticipantPermission{
|
|
CanPublish: true,
|
|
},
|
|
})
|
|
require.Error(t, err)
|
|
var twErr twirp.Error
|
|
require.True(t, errors.As(err, &twErr))
|
|
require.Equal(t, twirp.NotFound, twErr.Code())
|
|
})
|
|
}
|
|
|
|
// Ensure that CORS headers are returned
|
|
func TestSingleNodeCORS(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
s, finish := setupSingleNodeTest("TestSingleNodeCORS")
|
|
defer finish()
|
|
|
|
req, err := http.NewRequest("POST", fmt.Sprintf("http://localhost:%d", s.HTTPPort()), nil)
|
|
require.NoError(t, err)
|
|
req.Header.Set("Authorization", "bearer xyz")
|
|
req.Header.Set("Origin", "testhost.com")
|
|
res, err := http.DefaultClient.Do(req)
|
|
require.NoError(t, err)
|
|
require.Equal(t, "testhost.com", res.Header.Get("Access-Control-Allow-Origin"))
|
|
}
|
|
|
|
func TestSingleNodeDoubleSlash(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
s, finish := setupSingleNodeTest("TestSingleNodeDoubleSlash")
|
|
defer finish()
|
|
// client contains trailing slash in URL, causing path to contain double //
|
|
// without our middleware, this would cause a 302 redirect
|
|
roomClient = livekit.NewRoomServiceJSONClient(fmt.Sprintf("http://localhost:%d/", s.HTTPPort()), &http.Client{})
|
|
_, err := roomClient.ListRooms(contextWithToken(listRoomToken()), &livekit.ListRoomsRequest{})
|
|
require.NoError(t, err)
|
|
}
|
|
|
|
func TestPingPong(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
_, finish := setupSingleNodeTest("TestPingPong")
|
|
defer finish()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
c1 := createRTCClient("c1", defaultServerPort, testRTCServicePath, nil)
|
|
waitUntilConnected(t, c1)
|
|
|
|
require.NoError(t, c1.SendPing())
|
|
require.Eventually(t, func() bool {
|
|
return c1.PongReceivedAt() > 0
|
|
}, time.Second, 10*time.Millisecond)
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestSingleNodeJoinAfterClose(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
_, finish := setupSingleNodeTest("TestJoinAfterClose")
|
|
defer finish()
|
|
|
|
scenarioJoinClosedRoom(t)
|
|
}
|
|
|
|
func TestSingleNodeCloseNonRTCRoom(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
_, finish := setupSingleNodeTest("closeNonRTCRoom")
|
|
defer finish()
|
|
|
|
closeNonRTCRoom(t)
|
|
}
|
|
|
|
func TestAutoCreate(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
disableAutoCreate := func(conf *config.Config) {
|
|
conf.Room.AutoCreate = false
|
|
}
|
|
t.Run("cannot join if room isn't created", func(t *testing.T) {
|
|
s := createSingleNodeServer(disableAutoCreate)
|
|
go func() {
|
|
if err := s.Start(); err != nil {
|
|
logger.Errorw("server returned error", err)
|
|
}
|
|
}()
|
|
defer s.Stop(true)
|
|
|
|
waitForServerToStart(s)
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
token := joinToken(testRoom, "start-before-create", nil)
|
|
opts := &testclient.Options{}
|
|
testRTCServicePathToTestClientOptions(testRTCServicePath, opts)
|
|
_, err := testclient.NewWebSocketConn(
|
|
fmt.Sprintf("ws://localhost:%d", defaultServerPort),
|
|
token,
|
|
opts,
|
|
)
|
|
require.Error(t, err)
|
|
|
|
// second join should also fail
|
|
token = joinToken(testRoom, "start-before-create-2", nil)
|
|
_, err = testclient.NewWebSocketConn(
|
|
fmt.Sprintf("ws://localhost:%d", defaultServerPort),
|
|
token,
|
|
opts,
|
|
)
|
|
require.Error(t, err)
|
|
})
|
|
}
|
|
})
|
|
|
|
t.Run("join with explicit createRoom", func(t *testing.T) {
|
|
s := createSingleNodeServer(disableAutoCreate)
|
|
go func() {
|
|
if err := s.Start(); err != nil {
|
|
logger.Errorw("server returned error", err)
|
|
}
|
|
}()
|
|
defer s.Stop(true)
|
|
|
|
waitForServerToStart(s)
|
|
|
|
// explicitly create
|
|
_, err := roomClient.CreateRoom(contextWithToken(createRoomToken()), &livekit.CreateRoomRequest{Name: testRoom})
|
|
require.NoError(t, err)
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
c1 := createRTCClient("join-after-create", defaultServerPort, testRTCServicePath, nil)
|
|
waitUntilConnected(t, c1)
|
|
|
|
c1.Stop()
|
|
})
|
|
}
|
|
})
|
|
}
|
|
|
|
// don't give user subscribe permissions initially, and ensure autosubscribe is triggered afterwards
|
|
func TestSingleNodeUpdateSubscriptionPermissions(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
_, finish := setupSingleNodeTest("TestSingleNodeUpdateSubscriptionPermissions")
|
|
defer finish()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
pub := createRTCClient("pub", defaultServerPort, testRTCServicePath, nil)
|
|
|
|
grant := &auth.VideoGrant{RoomJoin: true, Room: testRoom}
|
|
grant.SetCanSubscribe(false)
|
|
at := auth.NewAccessToken(testApiKey, testApiSecret).
|
|
AddGrant(grant).
|
|
SetIdentity("sub")
|
|
token, err := at.ToJWT()
|
|
require.NoError(t, err)
|
|
sub := createRTCClientWithToken(token, defaultServerPort, testRTCServicePath, nil)
|
|
|
|
waitUntilConnected(t, pub, sub)
|
|
|
|
writers := publishTracksForClients(t, pub)
|
|
defer stopWriters(writers...)
|
|
|
|
// wait sub receives tracks
|
|
testutils.WithTimeout(t, func() string {
|
|
pubRemote := sub.GetRemoteParticipant(pub.ID())
|
|
if pubRemote == nil {
|
|
return "could not find remote publisher"
|
|
}
|
|
if len(pubRemote.Tracks) != 2 {
|
|
return "did not receive metadata for published tracks"
|
|
}
|
|
return ""
|
|
})
|
|
|
|
// set permissions out of band
|
|
ctx := contextWithToken(adminRoomToken(testRoom))
|
|
_, err = roomClient.UpdateParticipant(ctx, &livekit.UpdateParticipantRequest{
|
|
Room: testRoom,
|
|
Identity: "sub",
|
|
Permission: &livekit.ParticipantPermission{
|
|
CanSubscribe: true,
|
|
CanPublish: true,
|
|
},
|
|
})
|
|
require.NoError(t, err)
|
|
|
|
testutils.WithTimeout(t, func() string {
|
|
tracks := sub.SubscribedTracks()[pub.ID()]
|
|
if len(tracks) == 2 {
|
|
return ""
|
|
} else {
|
|
return fmt.Sprintf("expected 2 tracks subscribed, actual: %d", len(tracks))
|
|
}
|
|
})
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestSingleNodeAttributes(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
_, finish := setupSingleNodeTest("TestSingleNodeAttributes")
|
|
defer finish()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
pub := createRTCClient("pub", defaultServerPort, testRTCServicePath, &testclient.Options{
|
|
Attributes: map[string]string{
|
|
"b": "2",
|
|
"c": "3",
|
|
},
|
|
TokenCustomizer: func(token *auth.AccessToken, grants *auth.VideoGrant) {
|
|
T := true
|
|
grants.CanUpdateOwnMetadata = &T
|
|
token.SetAttributes(map[string]string{
|
|
"a": "0",
|
|
"b": "1",
|
|
})
|
|
},
|
|
})
|
|
|
|
grant := &auth.VideoGrant{RoomJoin: true, Room: testRoom}
|
|
grant.SetCanSubscribe(false)
|
|
at := auth.NewAccessToken(testApiKey, testApiSecret).
|
|
SetVideoGrant(grant).
|
|
SetIdentity("sub")
|
|
token, err := at.ToJWT()
|
|
require.NoError(t, err)
|
|
sub := createRTCClientWithToken(token, defaultServerPort, testRTCServicePath, nil)
|
|
|
|
waitUntilConnected(t, pub, sub)
|
|
|
|
// wait sub receives initial attributes
|
|
testutils.WithTimeout(t, func() string {
|
|
pubRemote := sub.GetRemoteParticipant(pub.ID())
|
|
if pubRemote == nil {
|
|
return "could not find remote publisher"
|
|
}
|
|
attrs := pubRemote.Attributes
|
|
if !reflect.DeepEqual(attrs, map[string]string{
|
|
"a": "0",
|
|
"b": "2",
|
|
"c": "3",
|
|
}) {
|
|
return fmt.Sprintf("did not receive expected attributes: %v", attrs)
|
|
}
|
|
return ""
|
|
})
|
|
})
|
|
}
|
|
}
|
|
|
|
// TestDeviceCodecOverride checks that codecs that are incompatible with a device is not
|
|
// negotiated by the server
|
|
func TestDeviceCodecOverride(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
_, finish := setupSingleNodeTest("TestDeviceCodecOverride")
|
|
defer finish()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
// simulate device that isn't compatible with H.264
|
|
c1 := createRTCClient("c1", defaultServerPort, testRTCServicePath, &testclient.Options{
|
|
ClientInfo: &livekit.ClientInfo{
|
|
Os: "android",
|
|
DeviceModel: "Xiaomi 2201117TI",
|
|
},
|
|
})
|
|
defer c1.Stop()
|
|
waitUntilConnected(t, c1)
|
|
|
|
// it doesn't really matter what the codec set here is, uses default Pion MediaEngine codecs
|
|
tw, err := c1.AddStaticTrack("video/h264", "video", "webcam")
|
|
require.NoError(t, err)
|
|
defer stopWriters(tw)
|
|
|
|
var desc *sdp.MediaDescription
|
|
require.Eventually(t, func() bool {
|
|
lastAnswer := c1.LastAnswer()
|
|
if lastAnswer == nil {
|
|
return false
|
|
}
|
|
|
|
sd := webrtc.SessionDescription{
|
|
Type: webrtc.SDPTypeAnswer,
|
|
SDP: lastAnswer.SDP,
|
|
}
|
|
answer, err := sd.Unmarshal()
|
|
require.NoError(t, err)
|
|
|
|
// video and data channel
|
|
if len(answer.MediaDescriptions) < 2 {
|
|
return false
|
|
}
|
|
|
|
for _, md := range answer.MediaDescriptions {
|
|
if md.MediaName.Media == "video" {
|
|
desc = md
|
|
break
|
|
}
|
|
}
|
|
return desc != nil
|
|
}, waitTimeout, waitTick, "did not receive answer")
|
|
|
|
hasSeenVP8 := false
|
|
for _, a := range desc.Attributes {
|
|
if a.Key == "rtpmap" {
|
|
require.NotContains(t, a.Value, mime.MimeTypeCodecH264.String(), "should not contain H264 codec")
|
|
if strings.Contains(a.Value, mime.MimeTypeCodecVP8.String()) {
|
|
hasSeenVP8 = true
|
|
}
|
|
}
|
|
}
|
|
require.True(t, hasSeenVP8, "should have seen VP8 codec in SDP")
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestSubscribeToCodecUnsupported(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
_, finish := setupSingleNodeTest("TestSubscribeToCodecUnsupported")
|
|
defer finish()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
c1 := createRTCClient("c1", defaultServerPort, testRTCServicePath, nil)
|
|
// create a client that doesn't support H264
|
|
c2 := createRTCClient("c2", defaultServerPort, testRTCServicePath, &testclient.Options{
|
|
AutoSubscribe: true,
|
|
DisabledCodecs: []webrtc.RTPCodecCapability{
|
|
{MimeType: "video/H264"},
|
|
},
|
|
})
|
|
waitUntilConnected(t, c1, c2)
|
|
|
|
// publish a vp8 video track and ensure c2 receives it ok
|
|
t1, err := c1.AddStaticTrack("audio/opus", "audio", "webcam")
|
|
require.NoError(t, err)
|
|
defer t1.Stop()
|
|
t2, err := c1.AddStaticTrack("video/vp8", "video", "webcam")
|
|
require.NoError(t, err)
|
|
defer t2.Stop()
|
|
|
|
testutils.WithTimeout(t, func() string {
|
|
if len(c2.SubscribedTracks()) == 0 {
|
|
return "c2 was not subscribed to anything"
|
|
}
|
|
// should have received two tracks
|
|
if len(c2.SubscribedTracks()[c1.ID()]) != 2 {
|
|
return "c2 was not subscribed to tracks from c1"
|
|
}
|
|
|
|
tracks := c2.SubscribedTracks()[c1.ID()]
|
|
for _, t := range tracks {
|
|
if mime.IsMimeTypeStringVP8(t.Codec().MimeType) {
|
|
return ""
|
|
}
|
|
}
|
|
return "did not receive track with vp8"
|
|
})
|
|
require.Nil(t, c2.GetSubscriptionResponseAndClear())
|
|
|
|
// publish a h264 track and ensure c2 got subscription error
|
|
t3, err := c1.AddStaticTrackWithCodec(webrtc.RTPCodecCapability{
|
|
MimeType: "video/h264",
|
|
ClockRate: 90000,
|
|
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
|
|
}, "videoscreen", "screen")
|
|
defer t3.Stop()
|
|
require.NoError(t, err)
|
|
|
|
var h264TrackID string
|
|
require.Eventually(t, func() bool {
|
|
remoteC1 := c2.GetRemoteParticipant(c1.ID())
|
|
require.NotNil(t, remoteC1)
|
|
for _, track := range remoteC1.Tracks {
|
|
if mime.IsMimeTypeStringH264(track.MimeType) {
|
|
h264TrackID = track.Sid
|
|
return true
|
|
}
|
|
}
|
|
return false
|
|
}, time.Second, 10*time.Millisecond, "did not receive track info with h264")
|
|
|
|
require.Eventually(t, func() bool {
|
|
sr := c2.GetSubscriptionResponseAndClear()
|
|
if sr == nil {
|
|
return false
|
|
}
|
|
require.Equal(t, h264TrackID, sr.TrackSid)
|
|
require.Equal(t, livekit.SubscriptionError_SE_CODEC_UNSUPPORTED, sr.Err)
|
|
return true
|
|
}, 5*time.Second, 10*time.Millisecond, "did not receive subscription response")
|
|
|
|
// publish another vp8 track again, ensure the transport recovered by sfu and c2 can receive it
|
|
t4, err := c1.AddStaticTrack("video/vp8", "video2", "webcam2")
|
|
require.NoError(t, err)
|
|
defer t4.Stop()
|
|
|
|
testutils.WithTimeout(t, func() string {
|
|
if len(c2.SubscribedTracks()) == 0 {
|
|
return "c2 was not subscribed to anything"
|
|
}
|
|
// should have received two tracks
|
|
if len(c2.SubscribedTracks()[c1.ID()]) != 3 {
|
|
return "c2 was not subscribed to tracks from c1"
|
|
}
|
|
|
|
var vp8Count int
|
|
tracks := c2.SubscribedTracks()[c1.ID()]
|
|
for _, t := range tracks {
|
|
if mime.IsMimeTypeStringVP8(t.Codec().MimeType) {
|
|
vp8Count++
|
|
}
|
|
}
|
|
if vp8Count == 2 {
|
|
return ""
|
|
}
|
|
return "did not 2 receive track with vp8"
|
|
})
|
|
require.Nil(t, c2.GetSubscriptionResponseAndClear())
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestDataPublishSlowSubscriber(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
dataChannelSlowThreshold := 21024
|
|
|
|
logger.Infow("----------------STARTING TEST----------------", "test", t.Name())
|
|
s := createSingleNodeServer(func(c *config.Config) {
|
|
c.RTC.DatachannelSlowThreshold = dataChannelSlowThreshold
|
|
})
|
|
go func() {
|
|
if err := s.Start(); err != nil {
|
|
logger.Errorw("server returned error", err)
|
|
}
|
|
}()
|
|
|
|
waitForServerToStart(s)
|
|
|
|
defer func() {
|
|
s.Stop(true)
|
|
logger.Infow("----------------FINISHING TEST----------------", "test", t.Name())
|
|
}()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
pub := createRTCClient("pub", defaultServerPort, testRTCServicePath, nil)
|
|
fastSub := createRTCClient("fastSub", defaultServerPort, testRTCServicePath, nil)
|
|
slowSubNotDrop := createRTCClient("slowSubNotDrop", defaultServerPort, testRTCServicePath, nil)
|
|
slowSubDrop := createRTCClient("slowSubDrop", defaultServerPort, testRTCServicePath, nil)
|
|
waitUntilConnected(t, pub, fastSub, slowSubDrop, slowSubNotDrop)
|
|
defer func() {
|
|
pub.Stop()
|
|
fastSub.Stop()
|
|
slowSubNotDrop.Stop()
|
|
slowSubDrop.Stop()
|
|
}()
|
|
|
|
// no data should be dropped for fast subscriber
|
|
var fastDataIndex atomic.Uint64
|
|
fastSub.OnDataReceived = func(data []byte, sid string) {
|
|
idx := binary.BigEndian.Uint64(data[len(data)-8:])
|
|
require.Equal(t, fastDataIndex.Load()+1, idx)
|
|
fastDataIndex.Store(idx)
|
|
}
|
|
|
|
// no data should be dropped for slow subscriber that is above threshold
|
|
var slowNoDropDataIndex atomic.Uint64
|
|
var drainSlowSubNotDrop atomic.Bool
|
|
slowNoDropReader := testclient.NewDataChannelReader(dataChannelSlowThreshold * 2)
|
|
slowSubNotDrop.OnDataReceived = func(data []byte, sid string) {
|
|
idx := binary.BigEndian.Uint64(data[len(data)-8:])
|
|
require.Equal(t, slowNoDropDataIndex.Load()+1, idx)
|
|
slowNoDropDataIndex.Store(idx)
|
|
if !drainSlowSubNotDrop.Load() {
|
|
slowNoDropReader.Read(data, sid)
|
|
}
|
|
}
|
|
|
|
// data should be dropped for slow subscriber that is below threshold
|
|
var slowDropDataIndex atomic.Uint64
|
|
dropped := make(chan struct{})
|
|
slowDropReader := testclient.NewDataChannelReader(dataChannelSlowThreshold / 2)
|
|
slowSubDrop.OnDataReceived = func(data []byte, sid string) {
|
|
select {
|
|
case <-dropped:
|
|
return
|
|
default:
|
|
}
|
|
idx := binary.BigEndian.Uint64(data[len(data)-8:])
|
|
if idx != slowDropDataIndex.Load()+1 {
|
|
close(dropped)
|
|
}
|
|
slowDropDataIndex.Store(idx)
|
|
slowDropReader.Read(data, sid)
|
|
}
|
|
|
|
// publisher sends data as fast as possible, it will block by the slowest subscriber above the slow threshold
|
|
var (
|
|
blocked atomic.Bool
|
|
stopWrite atomic.Bool
|
|
writeIdx atomic.Uint64
|
|
)
|
|
writeStopped := make(chan struct{})
|
|
go func() {
|
|
defer close(writeStopped)
|
|
var i int
|
|
buf := make([]byte, 100)
|
|
for !stopWrite.Load() {
|
|
i++
|
|
binary.BigEndian.PutUint64(buf[len(buf)-8:], uint64(i))
|
|
if err := pub.PublishData(buf, livekit.DataPacket_RELIABLE); err != nil {
|
|
if errors.Is(err, datachannel.ErrDataDroppedBySlowReader) {
|
|
blocked.Store(true)
|
|
i--
|
|
continue
|
|
} else {
|
|
t.Log("error writing", err)
|
|
break
|
|
}
|
|
}
|
|
writeIdx.Store(uint64(i))
|
|
}
|
|
}()
|
|
|
|
<-dropped
|
|
|
|
time.Sleep(time.Second)
|
|
blocked.Store(false)
|
|
require.Eventually(t, func() bool { return blocked.Load() }, 30*time.Second, 100*time.Millisecond)
|
|
stopWrite.Store(true)
|
|
<-writeStopped
|
|
drainSlowSubNotDrop.Store(true)
|
|
require.Eventually(t, func() bool {
|
|
return writeIdx.Load() == fastDataIndex.Load() &&
|
|
writeIdx.Load() == slowNoDropDataIndex.Load()
|
|
}, 10*time.Second, 50*time.Millisecond, "writeIdx %d, fast %d, slowNoDrop %d", writeIdx.Load(), fastDataIndex.Load(), slowNoDropDataIndex.Load())
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestFireTrackBySdp(t *testing.T) {
|
|
_, finish := setupSingleNodeTest("TestFireTrackBySdp")
|
|
defer finish()
|
|
|
|
var cases = []struct {
|
|
name string
|
|
codecs []webrtc.RTPCodecCapability
|
|
pubSDK livekit.ClientInfo_SDK
|
|
}{
|
|
{
|
|
name: "js client could pub a/v tracks",
|
|
codecs: []webrtc.RTPCodecCapability{
|
|
{MimeType: mime.MimeTypeH264.String()},
|
|
{MimeType: mime.MimeTypeOpus.String()},
|
|
},
|
|
pubSDK: livekit.ClientInfo_JS,
|
|
},
|
|
{
|
|
name: "go client could pub audio tracks",
|
|
codecs: []webrtc.RTPCodecCapability{
|
|
{MimeType: "audio/opus"},
|
|
},
|
|
pubSDK: livekit.ClientInfo_GO,
|
|
},
|
|
}
|
|
|
|
for _, c := range cases {
|
|
codecs, sdk := c.codecs, c.pubSDK
|
|
t.Run(c.name, func(t *testing.T) {
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
c1 := createRTCClient(c.name+"_c1", defaultServerPort, testRTCServicePath, &testclient.Options{
|
|
ClientInfo: &livekit.ClientInfo{
|
|
Sdk: sdk,
|
|
},
|
|
})
|
|
c2 := createRTCClient(c.name+"_c2", defaultServerPort, testRTCServicePath, &testclient.Options{
|
|
AutoSubscribe: true,
|
|
ClientInfo: &livekit.ClientInfo{
|
|
Sdk: livekit.ClientInfo_JS,
|
|
},
|
|
})
|
|
waitUntilConnected(t, c1, c2)
|
|
defer func() {
|
|
c1.Stop()
|
|
c2.Stop()
|
|
}()
|
|
|
|
// publish tracks and don't write any packets
|
|
for _, codec := range codecs {
|
|
_, err := c1.AddStaticTrackWithCodec(codec, codec.MimeType, codec.MimeType, testclient.AddTrackNoWriter())
|
|
require.NoError(t, err)
|
|
}
|
|
|
|
require.Eventually(t, func() bool {
|
|
return len(c2.SubscribedTracks()[c1.ID()]) == len(codecs)
|
|
}, 5*time.Second, 10*time.Millisecond)
|
|
|
|
var found int
|
|
for _, pubTrack := range c1.GetPublishedTrackIDs() {
|
|
t.Log("pub track", pubTrack)
|
|
tracks := c2.SubscribedTracks()[c1.ID()]
|
|
for _, track := range tracks {
|
|
t.Log("sub track", track.ID(), track.Codec())
|
|
if track.Codec().PayloadType == 0 && track.ID() == pubTrack {
|
|
found++
|
|
break
|
|
}
|
|
}
|
|
}
|
|
require.Equal(t, len(codecs), found)
|
|
})
|
|
}
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestSinglePublisherDataTrack(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
s, finish := setupSingleNodeTest("TestSinglePublisherDataTrack")
|
|
defer finish()
|
|
|
|
for _, testRTCServicePath := range testRTCServicePaths {
|
|
t.Run(fmt.Sprintf("testRTCServicePath=%s", testRTCServicePath.String()), func(t *testing.T) {
|
|
c1 := createRTCClient("c1", defaultServerPort, testRTCServicePath, &testclient.Options{AutoSubscribeDataTrack: true})
|
|
c2 := createRTCClient("c2", defaultServerPort, testRTCServicePath, &testclient.Options{AutoSubscribeDataTrack: true})
|
|
waitUntilConnected(t, c1, c2)
|
|
|
|
// publish a couple of data tracks and ensure clients receive it ok
|
|
dt1, err := c1.PublishDataTrack()
|
|
require.NoError(t, err)
|
|
defer dt1.Stop()
|
|
|
|
dt2, err := c1.PublishDataTrack()
|
|
require.NoError(t, err)
|
|
defer dt2.Stop()
|
|
|
|
testutils.WithTimeout(t, func() string {
|
|
if len(c2.SubscribedDataTracks()) == 0 {
|
|
return "c2 was not subscribed to any data tracks"
|
|
}
|
|
// should have received two data tracks
|
|
if len(c2.SubscribedDataTracks()[c1.ID()]) != 2 {
|
|
return "c2 didn't subscribe to both data tracks from c1"
|
|
}
|
|
return ""
|
|
})
|
|
|
|
// a new client joins and should get the initial stream
|
|
c3 := createRTCClient("c3", defaultServerPort, testRTCServicePath, &testclient.Options{AutoSubscribeDataTrack: true})
|
|
|
|
// ensure that new client that has joined also received data tracks
|
|
waitUntilConnected(t, c3)
|
|
testutils.WithTimeout(t, func() string {
|
|
if len(c3.SubscribedDataTracks()) == 0 {
|
|
return "c3 didn't subscribe to any data tracks"
|
|
}
|
|
// should have received two data tracks
|
|
if len(c3.SubscribedDataTracks()[c1.ID()]) != 2 {
|
|
return "c3 didn't subscribe to tracks from c1"
|
|
}
|
|
return ""
|
|
})
|
|
|
|
// ensure that the data track ids are generated by server
|
|
tracks := c3.SubscribedDataTracks()[c1.ID()]
|
|
for _, tr := range tracks {
|
|
require.True(t, strings.HasPrefix(string(tr.ID()), "DTR_"), "data track should begin with DTR")
|
|
}
|
|
|
|
// when c3 disconnects, ensure subscriber is cleaned up correctly
|
|
c3.Stop()
|
|
|
|
testutils.WithTimeout(t, func() string {
|
|
room := s.RoomManager().GetRoom(context.Background(), testRoom)
|
|
p := room.GetParticipant("c1")
|
|
require.NotNil(t, p)
|
|
|
|
for _, t := range p.GetPublishedDataTracks() {
|
|
if t.IsSubscriber(c3.ID()) {
|
|
return "c3 was not a subscriber of c1's data tracks"
|
|
}
|
|
}
|
|
return ""
|
|
})
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestTurnRelay(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
testCases := []struct {
|
|
name string
|
|
allowRestrictedPeerCIDRs []string
|
|
denyPeerCIDRs []string
|
|
expectedToConnect bool
|
|
}{
|
|
{
|
|
"allow",
|
|
[]string{"10.0.0.0/8", "192.168.0.0/16"},
|
|
nil,
|
|
true,
|
|
},
|
|
{
|
|
"not-allowed",
|
|
nil,
|
|
nil,
|
|
false,
|
|
},
|
|
{
|
|
"denied-overrides-allowed",
|
|
[]string{"10.0.0.0/8", "192.168.0.0/16"},
|
|
[]string{"10.0.0.0/8", "192.168.0.0/16"},
|
|
false,
|
|
},
|
|
}
|
|
|
|
for _, tc := range testCases {
|
|
t.Run(tc.name, func(t *testing.T) {
|
|
s := createSingleNodeServer(func(c *config.Config) {
|
|
c.TURN.Enabled = true
|
|
c.TURN.UDPPort = 3478
|
|
c.TURN.AllowRestrictedPeerCIDRs = tc.allowRestrictedPeerCIDRs
|
|
c.TURN.DenyPeerCIDRs = tc.denyPeerCIDRs
|
|
})
|
|
go func() {
|
|
if err := s.Start(); err != nil {
|
|
logger.Errorw("server returned error", err)
|
|
}
|
|
}()
|
|
defer s.Stop(true)
|
|
|
|
waitForServerToStart(s)
|
|
|
|
c1 := createRTCClient("relay_c1", defaultServerPort, testRTCServicePathv0, &testclient.Options{
|
|
AutoSubscribe: true,
|
|
ForceRelay: true,
|
|
})
|
|
defer c1.Stop()
|
|
|
|
if tc.expectedToConnect {
|
|
waitUntilConnected(t, c1)
|
|
|
|
testutils.WithTimeout(t, func() string {
|
|
if !c1.IsLocalCandidateRelaySelected() {
|
|
return "expected local candidate to be relay"
|
|
}
|
|
return ""
|
|
})
|
|
} else {
|
|
ensureNotConnected(t, c1)
|
|
}
|
|
})
|
|
}
|
|
}
|
|
|
|
func TestTurnAuthFailure(t *testing.T) {
|
|
if testing.Short() {
|
|
t.SkipNow()
|
|
return
|
|
}
|
|
|
|
const turnUDPPort = 3478
|
|
|
|
s := createSingleNodeServer(func(c *config.Config) {
|
|
c.TURN.Enabled = true
|
|
c.TURN.UDPPort = turnUDPPort
|
|
})
|
|
go func() {
|
|
if err := s.Start(); err != nil {
|
|
logger.Errorw("server returned error", err)
|
|
}
|
|
}()
|
|
defer s.Stop(true)
|
|
|
|
waitForServerToStart(s)
|
|
|
|
// build a known-good username/password pair so individual cases can mutate
|
|
// only the part they are exercising.
|
|
pID := livekit.ParticipantID("PA_authfail")
|
|
authHandler := service.NewTURNAuthHandler(auth.NewSimpleKeyProvider(testApiKey, testApiSecret))
|
|
validUsername, validExpiry := authHandler.CreateUsername(testApiKey, pID, 300)
|
|
validPassword, err := authHandler.CreatePassword(testApiKey, pID, validExpiry)
|
|
require.NoError(t, err)
|
|
|
|
// username encoded with an already-expired timestamp.
|
|
expiredUsername, _ := authHandler.CreateUsername(testApiKey, pID, -10)
|
|
|
|
// username encoded with an api key the server does not know about.
|
|
unknownAPIKeyUsername, _ := authHandler.CreateUsername("unknown-api-key", pID, 300)
|
|
|
|
// password whose hash was generated for an expiry that doesn't match the
|
|
// one encoded in the username. The server reconstructs the password using
|
|
// the username's expiry, so the integrity check fails.
|
|
mismatchedExpiryPassword, err := authHandler.CreatePassword(testApiKey, pID, validExpiry+60)
|
|
require.NoError(t, err)
|
|
require.NotEqual(t, validPassword, mismatchedExpiryPassword)
|
|
|
|
// username carrying expiry=0 must be rejected outright; constructed
|
|
// directly because CreateUsername always stamps a real expiry.
|
|
zeroExpiryUsername := base62.EncodeToString(fmt.Appendf(nil, "%s|%s|%d", testApiKey, pID, 0))
|
|
|
|
// username with only apiKey|pID (no expiry component) is the legacy
|
|
// pre-expiry form and must be rejected.
|
|
twoPartUsername := base62.EncodeToString(fmt.Appendf(nil, "%s|%s", testApiKey, pID))
|
|
|
|
testCases := []struct {
|
|
name string
|
|
username string
|
|
password string
|
|
}{
|
|
{
|
|
name: "unparseable-username",
|
|
username: "not-base62!!!",
|
|
password: validPassword,
|
|
},
|
|
{
|
|
name: "wrong-password",
|
|
username: validUsername,
|
|
password: "wrongpassword",
|
|
},
|
|
{
|
|
name: "expired-username",
|
|
username: expiredUsername,
|
|
password: validPassword,
|
|
},
|
|
{
|
|
name: "unknown-api-key",
|
|
username: unknownAPIKeyUsername,
|
|
password: validPassword,
|
|
},
|
|
{
|
|
name: "password-expiry-mismatch",
|
|
username: validUsername,
|
|
password: mismatchedExpiryPassword,
|
|
},
|
|
{
|
|
name: "zero-expiry-username",
|
|
username: zeroExpiryUsername,
|
|
password: validPassword,
|
|
},
|
|
{
|
|
name: "two-part-username",
|
|
username: twoPartUsername,
|
|
password: validPassword,
|
|
},
|
|
}
|
|
|
|
for _, tc := range testCases {
|
|
t.Run(tc.name, func(t *testing.T) {
|
|
conn, err := net.ListenPacket("udp4", "0.0.0.0:0")
|
|
require.NoError(t, err)
|
|
defer conn.Close()
|
|
|
|
client, err := turn.NewClient(&turn.ClientConfig{
|
|
TURNServerAddr: fmt.Sprintf("127.0.0.1:%d", turnUDPPort),
|
|
Username: tc.username,
|
|
Password: tc.password,
|
|
Realm: service.LivekitRealm,
|
|
Conn: conn,
|
|
})
|
|
require.NoError(t, err)
|
|
defer client.Close()
|
|
require.NoError(t, client.Listen())
|
|
|
|
_, allocErr := client.Allocate()
|
|
require.Error(t, allocErr)
|
|
|
|
// pion's TURN server replies with 400 Bad Request for any
|
|
// authenticated-allocate failure (unknown user or integrity check
|
|
// mismatch); the initial unauthenticated probe is what returns 401.
|
|
var turnErr *stun.TurnError
|
|
require.ErrorAs(t, allocErr, &turnErr)
|
|
require.Equal(t, stun.CodeBadRequest, turnErr.ErrorCodeAttr.Code)
|
|
})
|
|
}
|
|
}
|