863ba053ef
Audio rework, new jitter buffer
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Jitter buffer is now based on packets, not on samples. The frames are
dejittered in received form. After reading from jitter buffer, they are
decoded in correct order. If a frame is missing, it is concealed by
repeating audio.
2024-03-25 21:30:31 +01:00
14b2df8907
Add parameter to control increase speed of compressor
2024-03-25 21:29:49 +01:00
5b96cf935b
Change IE_METERING timing information from deciseconds to seconds and milliseconds
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Parameter "metering" now accepts parameter "period" in seconds as floating point value and gets converted to seconds and milliseconds.
2024-01-29 11:36:50 +01:00
Dennis Grunert
d8ebec0d6f
Provide CC metering information to endpoints
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A new routing command 'metering' has been added. It has two parameters to specify metering information:
- connect_units (units charged upon answer, normally '1'; set to '0' to mark the call as free of charge; set to higher value to add a call-setup charge)
- period_decisecs (time period of one unit in 1/10 seconds; set to '0' for a call-setup charge only)
Metering details are encoded within IE_METERING and attached to the next outgoing CC-PROC-REQ/CC-ALERT-REQ/CC-SETUP-RSP message for the call originator.
2024-01-25 20:09:12 +01:00
02510a9973
Move from local to external osmo* libraries
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src/libdebug -> libosmocore
src/libselect -> libosmocore
src/libtimer -> libosmocore
src/libosmocc -> libosmo-cc
src/libg711 -> libosmo-cc
2024-01-25 20:09:11 +01:00
212aac461d
If audio is available after disconnect, send SDP to origin, if not already
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Especially if disconnect is recevied prior connect. In this case we want
audio after alerting and not just silence.
2024-01-07 18:11:37 +01:00
16f5ae06f3
If interface is not connected, collect cause 27 and release is no interface
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Release call to one or more interface, if none of the interfaces are
available.
2023-12-10 14:43:00 +01:00
9ea1e4f10e
Add "sending-complete" to call function
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Used to indicate that number is complete. Some exchanges will accept #
as last digit too.
2023-12-01 17:34:35 +01:00
a138935937
Add support for telephone events
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Not much tested yet, except for sending telephone events via SIP.
2023-11-12 17:45:29 +01:00
59215ae370
Add compressor to maintain speech volume at normal speech level.
2023-06-17 21:45:34 +02:00
8546453296
fixup 675bada618
2023-01-23 18:44:06 +01:00
6bfa342aef
Moved from polling to select
2023-01-22 10:07:34 +01:00
675bada618
Correctly handle call parameters with value set to 0
2023-01-22 09:01:53 +01:00
fd0337b227
Fixed parameter parsing
2022-12-16 12:45:48 +01:00
137c50a33f
Add GSM (libgsm) codec support
2022-11-25 09:29:03 +01:00
20621011be
Fixes for telephone event messages
2022-10-30 17:04:42 +01:00
bbc4af49e9
Improved 'help' output
2022-10-30 17:04:39 +01:00
294f7e9d1c
Updated to new dejitter API
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Also moved jitter buffers, playback and recording to call instance, where they
belong to.
2022-10-30 17:04:36 +01:00
0adc855d96
minor description fix
2022-06-25 13:03:09 +02:00
67d66e73d2
Structured the commands and environment variables
2021-09-17 16:33:29 +02:00
42f4d661a9
Support for two endpoints. Useful for multi stack setups.
2021-09-17 16:33:29 +02:00
80d911e5a6
Updated libs
2021-09-17 16:33:29 +02:00
1625510377
Fixed cause parameter
2021-03-20 11:38:12 +01:00
d5face6404
Add DTMF detection via telephone-event payload
2021-03-14 11:20:55 +01:00
Martin Hauke
027846d895
Fix typos
2021-01-03 10:10:47 +01:00
fde7cc2ce3
Initial GIT import
2020-12-29 19:02:56 +01:00
Andreas Eversberg
Dennis Grunert
Martin Hauke