composer/projects/gk7205v200_ultimate_baresip/general/overlay/etc/baresip/config

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# Core
poll_method epoll # poll, select, epoll ..
# SIP
#sip_listen 0.0.0.0:5060
#sip_certificate cert.pem
sip_cafile /etc/ssl/certs/ca-certificates.crt
#sip_transports udp,tcp,tls,ws,wss
#sip_trans_def udp
#sip_verify_server yes
sip_tos 160
# Call
call_local_timeout 120
call_max_calls 1
call_hold_other_calls yes
# Audio
#audio_path /usr/share/baresip
audio_player goke
audio_source rtspausrc,localhost
#audio_alert alsa,default
#ausrc_srate 48000
#auplay_srate 48000
#ausrc_channels 0
#auplay_channels 0
#audio_txmode poll # poll, thread
audio_level no
ausrc_format s16 # s16, float, ..
auplay_format s16 # s16, float, ..
auenc_format s16 # s16, float, ..
audec_format s16 # s16, float, ..
audio_buffer 20-160 # ms
audio_telev_pt 101 # payload type for telephone-event
# Video
#video_source v4l2,/dev/video0
#video_source fakevideo,nil
#video_display x11,nil
video_size 640x480
video_bitrate 1000000
video_fps 30.00
video_fullscreen no
videnc_format yuv420p
# AVT - Audio/Video Transport
rtp_tos 184
rtp_video_tos 136
#rtp_ports 10000-20000
#rtp_bandwidth 512-1024 # [kbit/s]
rtcp_mux no
jitter_buffer_type fixed # off, fixed, adaptive
jitter_buffer_delay 5-10 # frames
#jitter_buffer_wish 6 # frames for start
rtp_stats no
rtp_timeout 4
module_path /usr/lib/baresip/modules
# UI Modules
module stdio.so
module g711.so
module webrtc_aec.so
module rtspausrc.so
module goke.so
#module fakevideo.so
#module avformat.so
module_app account.so
module_app contact.so
module_app debug_cmd.so
module_app syslog.so
module_app ctrl_tcp.so
#
module_app menu.so
#module_app mwi.so
#module_app presence.so
#module_app serreg.so
module_app netroam.so
# UI Modules parameters
cons_listen 0.0.0.0:3000 # cons - Console UI UDP/TCP sockets
ctrl_tcp_listen 0.0.0.0:2000 # ctrl_tcp - TCP interface JSON
evdev_device /dev/input/event0
# Opus codec parameters
opus_bitrate 28000 # 6000-510000
vumeter_stderr yes
# Selfview
video_selfview window # {window,pip}
#selfview_size 64x64