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meetecho
/
janus-gateway
Janus WebRTC Server
https://janus.conf.meetecho.com
GNU General Public License v3.0
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[1.x] why is it recommended to use libnice v0.1.18? libnice 0.1.19 exists a number of bug fixes
#3385
alexber220
closed
4 months ago
1
Bring back switches "-V" and "--version".
#3384
fancycode
closed
4 months ago
1
[1.x] Commandline option "--version" no longer exists
#3383
fancycode
closed
4 months ago
1
[1.x] Im used Janus in docker contaner with docker-compose
#3382
muramidaza
closed
4 months ago
1
Options to block startup unless RabbitMQ Transport / Event Handler are connected
#3381
chriswiggins
opened
4 months ago
4
AudioBridge: (re) configure remote rtp
#3380
keremcadirci
opened
4 months ago
5
Failed to compile janus on an arm machine inside docker container with ubuntu 24.04 as image
#3379
shivanshtalwar0
closed
4 months ago
1
Pull out all helper methods
#3378
redexp
opened
4 months ago
5
Added UMD
#3377
redexp
closed
3 months ago
11
[1.x] AudioBridge Talking Event is not firing
#3376
mail2mhossain
closed
4 months ago
8
Updated link to minijanus.js and janus-plugin-rs to maintained forks
#3375
vincentfretin
closed
4 months ago
1
Add note about freeing transaction in handle_message
#3374
vincentfretin
closed
4 months ago
6
Fixed memory leak in janus_sdp_get_codec_pt_full (master branch)
#3373
vincentfretin
closed
4 months ago
0
[1.x] Transport Websockets plugin binds to ANY network interface
#3372
YtelDrew
closed
4 months ago
4
Fixed memory leak in janus_sdp_get_codec_pt_full (0.x branch)
#3371
vincentfretin
closed
4 months ago
8
[1.x] SIP plugin "Couldn't connect audio RTP? -- 22 (Invalid argument)"
#3370
Pineappluh
closed
4 months ago
5
video room plugin: introduced string_ids_user
#3369
IgorKhomenko
closed
3 months ago
6
audiobridge: cleanup participant's queue and jitter buffer when muted…
#3368
atoppi
closed
3 weeks ago
0
[0.x] Possible memory leaks found with libasan latest
#3367
arpu
closed
4 months ago
12
[1.x] build failure on macos
#3366
abhijeetbhagat
closed
4 months ago
3
[1.x] Increased event handlers SIP logging buffer size one more time or make it configurable
#3365
ycherniavskyi
closed
2 months ago
7
[1.x] videoroom: support of string_ids/string_ids_user
#3364
IgorKhomenko
opened
5 months ago
2
[1.x]inconsistency happen when more than 6 users join at same time with different browsers (web and mobile)
#3363
venkateshcontus
closed
3 months ago
11
[1.x] inconsistency happen when more than 6 users join at same time
#3362
venkateshcontus
closed
5 months ago
2
Add new participant mutex to VideoRoom
#3361
lminiero
closed
4 months ago
1
[1.x] Crash / segfault on 1.2.3
#3360
tespio
closed
4 months ago
10
videoroom: Release some references created by remote publishers
#3359
fancycode
opened
5 months ago
3
[1.x] system under heavy load stops to handle calls wia ws api
#3358
spscream
closed
3 months ago
22
Add optional RNNoise support to AudioBridge (0.x)
#3357
lminiero
closed
5 months ago
0
Fix potential audiobridge crash when changing room (fixes #3327)
#3356
atoppi
closed
5 months ago
0
videoroom: keep subscriber reference for the whole message handler
#3355
atoppi
closed
5 months ago
0
[1.x] Crash / segfault on 1.2.2
#3354
tespio
closed
5 months ago
5
[0.x] [audiobridge] jitter buffer: manually update internal delay and set a max size
#3353
atoppi
closed
5 months ago
0
[1.x] rtp_port_range not being respected
#3352
colinturner99
closed
5 months ago
2
Commit of change ufrag based on what total_length
#3351
mayukhdev
closed
5 months ago
2
[1.x] Does AudioBridge supports multiple participant join from RTP in a single room . Getting `Already in a room` Error for 2nd participant
#3350
keremcadirci
closed
5 months ago
3
Add packet concealment for opus decoder in audiobridge
#3349
spscream
closed
3 months ago
14
Ship speexdsp's jitter buffer as part of local AudioBridge dependencies
#3348
lminiero
closed
5 months ago
6
Avoid applying an offset to a null pointer
#3347
atoppi
closed
6 months ago
1
Remove extra reference to remote publisher (fixes #3345)
#3346
atoppi
closed
6 months ago
0
[1.x] videoroom: remote publisher doesn't release RTP/RTCP ports
#3345
boddob
closed
6 months ago
10
[1.x] Issue with outbound calls for SIP Gateway
#3344
Magmanat
closed
6 months ago
3
janus.js: "connectionState" callback added
#3343
RSATom
closed
6 months ago
3
[0.x] janus.js: "connectionState" callback added
#3342
RSATom
closed
6 months ago
1
[0.x] [janus.js] black screen or streaming stops periodically when connected through WebRTC (Chromium only)
#3341
plia7
closed
6 months ago
4
Fix getBitrate()
#3340
avenal
closed
5 months ago
2
Don't call close_pc in SIP plugin if there was no SDP (fixes #3332)
#3339
lminiero
closed
6 months ago
1
Build fails with latest libsrtp version
#3338
hboetes
closed
7 months ago
9
Fixed broken createOffer documentation
#3337
mabolo
closed
6 months ago
2
Add unauthenticated no-payload ping endpoint for health checks
#3336
husam212
closed
7 months ago
3
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