7392d01d35
Reduced timeout of incoming calls to 60 sec
2026-05-05 15:28:19 +03:00
e247a57382
Added Unique Contact URI setting
2026-03-11 03:24:49 +02:00
38b5bef715
Added Check Origin account setting
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Get Settings and Account strings outside of composables
2025-12-25 11:25:12 +02:00
288679c6c0
Removed unused static config line
2025-10-08 16:35:52 +03:00
d0ce112a2a
Renamed cuser_random setting
2025-09-26 07:58:24 +03:00
54f27a9940
Removed commented out code (not needed anymore)
2025-09-02 15:00:22 +03:00
4c7f2d9a77
Reduced size of audio_buffer
2025-08-21 16:15:56 +03:00
55f4c6332e
Upstream audio_jitter_buffer changes
2025-08-21 16:08:39 +03:00
fcfdb1027a
Fixed telephone-event payload type number
2025-01-01 00:36:38 +02:00
71f7578705
Use aaudio instead of opensles
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Added audio effects ro recorder stream
2024-11-16 05:54:38 +02:00
f1a293b6d4
Changed The Test Call contact URI
2024-10-15 11:26:01 +03:00
0b01a4ad58
Revert "No need to check webrtc_aecm module when config is initialized"
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This reverts commit adfd15e1330c2040844e3c03d54502ea553187d5.
2024-01-12 03:29:47 +02:00
adfd15e133
No need to check webrtc_aecm module when config is initialized
2024-01-12 03:01:41 +02:00
0a0a2356b8
Use thread rx_mode
2024-01-12 02:51:20 +02:00
8b366493f3
Remove rtp tos values from config (use defaults)
2023-05-27 10:12:13 +03:00
e9ecfdd648
Major rewrite of settings implementation by separating static and dynamic
...
settings
Changed opus expected packed loss default value from 0 to 1 in order to enable
forward error correction
Restored original audio jitten buffer size to 0-20 now when upstream bug
is fixed
2023-05-25 14:36:45 +03:00
a295c43329
Make 1-50 change also in config file
2023-05-21 17:04:02 +03:00
3e408b9536
Revert back to original jitter buffer delays now when libre bug is fixed
2023-05-19 20:57:00 +03:00
68700a6036
Fixed jitter buffer delays
2023-05-18 22:09:28 +03:00
df76f5fbda
Terminate call if incoming rtp packets have not been received in 60 sec
2023-05-18 11:35:36 +03:00
a0a786c485
Separate audio and video jitter buffer configuration
2023-05-16 15:05:27 +03:00
a8029b18f1
Use audio_buffer 20-300 and jitter_buffer_delay 0-20
2023-05-10 18:42:06 +03:00
1528bf38ee
Use default values for audio_buffer and jitter_buffer_delay
2023-05-09 11:56:14 +03:00
111ae9a3ad
Removed unused contact module
2023-04-03 09:44:58 +03:00
292dbd275d
Removed rtcp_mux config variable
2022-12-12 03:33:11 +02:00
1eaa0935f2
Replaced zrtp module with gzrtp module
2022-10-07 20:04:04 +03:00
d859e9c220
Added gsm module
2022-09-02 11:58:32 +03:00
49d4b8f759
Removed obsolate ice config variables
2022-05-26 15:33:49 +03:00
ba8c44bc12
Use elliptic curve cryptography based DTLS Key Establishment Protocol
2022-04-22 11:25:51 +03:00
ba700b010e
Audio related config updates
2022-04-08 10:01:00 +03:00
ef825b57e6
Activate rtp_stats
2021-12-19 10:07:18 +02:00
af99daa245
No need for presence module
2021-12-09 14:20:19 +02:00
f0523e0fb0
Call transfer modifications
2021-12-08 10:03:58 +02:00
e35b0f6fb7
Removed webrtc_aec_extended_filter config parameter
2021-11-02 09:43:17 +02:00
2fa07d5421
Use BaresipService to play all sounds
2021-10-31 17:32:54 +02:00
3820fa6934
Replaced webrtc_aec module with webrtc_aecm module
2021-10-12 13:45:51 +03:00
72ccb8a3b1
Replaced deprecated tmp modules
2021-09-19 09:12:45 +03:00
3bf70f52b8
Restored accidently removed ringback asset
2021-08-31 22:07:04 +03:00
34348e4d37
Remove ilbc audio module (removed from upstream)
2021-06-03 18:42:53 +03:00
bf0b9f125a
Deleted unused audio assets
2021-06-03 10:21:32 +03:00
7bf2447287
Added 'Verify Server Certificates' setting
2021-02-28 15:48:01 +02:00
4a58ddbbe2
Use adaptive jitter buffer
2020-12-08 15:56:52 +02:00
7799dbeefb
Removed obsolete config vars audio_srate and audio_channels
2020-11-29 16:08:18 +02:00
6e8cc86199
Added G.729 audio codec
2020-11-26 11:03:24 +02:00
83f311f21c
- Moved audio settings to its own activity.
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- Added AEC Extented Filter audio setting.
2020-07-06 17:46:25 +03:00
ad7edeaf9f
- Removed ICE lite mode setting due to upstream removal.
2020-04-19 16:28:46 +03:00
86b60fa0f8
- Replaced global setting "Prefer IPv6" with account specific
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"Prefer IPv6 Media" setting.
- Improved acting on changes in network connectivity.
2020-02-07 01:19:41 +02:00
91717628f9
- Use net_set_address API function to set IPv6 address.
2019-11-03 11:46:17 +02:00
53f751454d
- Added to Settings possibility to choose which audio modules are loaded.
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Codecs provided by loaded modules are available for accounts to use.
2019-10-22 17:33:46 +03:00
06a18835b4
- Added AMR narrowband codec.
2019-10-14 11:19:01 +03:00
