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this applies to config values such as "audio_player",
"audio_source", "video_source" etc.
example:
audio_player alsa,"ALC887-VD Digital [ALC887-VD Digital]"
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contributed by Tomasz Ostrowski from the
"VoIP/SIP client (softphone) for Windows" project:
http://tomeko.net/software/SIPclient/
Review: alfredh
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thanks to Tomasz Ostrowski for reporting
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Opus config
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Merge upstream
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audio: fix audio_srate range for G.722
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Merge Upstream
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Upstream
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most of this work was done by Juha Heinanen, and reviewed by me.
thanks!
- menu: add support for sending SIP OPTIONS requests to any peer,
using the 'o' command
- presence: add support for PUBLISH, set ;pubint=N to activate
- ua: add Allow/Supported header to OPTIONS reply
add MESSAGE to allowed methods
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gruu: preliminary support for SIP GRUU
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- the original patch was written by Juha Heinanen
- this patch needs a new version of libre
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- change from multiple-instances to single-instance
(multiple-instances was never used)
- remove ui/input config parameters from the "core" config,
moved to each specific module (cons and evdev)
- modules updated: stdio, cons, evdev, wincons
- this patch solves the reported crash, that module-functions
are called after the module was unloaded.
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the python-program ccheck.py can be used to verify that the source
code does not contain any whitespace errors:
https://github.com/alfredh/pytools/blob/master/ccheck.py
how to use it:
$ cd baresip
$ ccheck.py
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- metric_init calls tmr_start(). tmr_start must be called from the
re_main() thread.
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contributed by Ingo Feinerer, thanks !
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- the patch was contributed by Ingo Feinerer
- use the command 'Z' to write verified ZID
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- tmr_start() should not be called from a thread that is not running
the re_main() loop. i.e. tmr_start() MUST be called from the re_main()
loop thread, unless the call is wrapped by re_thread_enter/leave().
- the audio-processing handlers encode/decode are handling real-time
audio samples, and should not be blocked.
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config: generate more relevant config for FreeBSD and OpenBSD
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- also minor cleanups
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- use enum AVCodecID instead of enum CodecID
- use AV_CODEC_ID_xxx instead of CODEC_ID_xxx
- extend a backwards compatible macro-wrapper in
avcodec.h
avcodec should still compile on old versions of FFmpeg.
please let us know if it does not :)
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- udp_send() will process *all* send helpers on that UDP-socket, from
the top-level helper to the bottom-level helper (including the
helper used by zrtp).
- instead we use udp_send_helper() which will send the outgoing packet
from the helper right *below* this helper (st->uh), and continue down
the chain of helpers.
- the logic behind UDP-helpers is that a particular helper should not
know about any other potential helpers above or below
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the patch was contributed by Ingo Feinerer, thanks!
- add support for generating a random ZID the first time
the application is started, and saving the ZID in
$HOME/.baresip/zrtp_zid
- subsequent runs will read the ZID from this file
- setting client_id to "baresip/zrtp"
- setting license-mode to UNLIMITED
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thanks to Remigiusz Borek for reporting the issue
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Credits to Francesco Bradascio for the original work.
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