Run baresip in a container and play an audio file, but the callee cannot hear the audio. #21
Description
I'm running the official baresip image and want to play an audio file during calls, but the callee cannot hear the audio. I'm using a MacBook Air M3 and running it with podman.
Below are the modifications I made based on the generated configuration:
#
# baresip configuration
#
#------------------------------------------------------------------------------
# SIP
#sip_listen 0.0.0.0:5060
#sip_certificate cert.pem
sip_cafile /etc/ssl/certs/ca-certificates.crt
sip_capath /etc/ssl/certs
#sip_transports udp,tcp,tls,ws,wss
#sip_trans_def udp
#sip_verify_server yes
#sip_verify_client no
#sip_tls_resumption all
sip_tos 160
#filter_registrar udp,tcp,tls,ws,wss
# Call
call_local_timeout 120
call_max_calls 4
call_hold_other_calls yes
call_accept no
# Audio
audio_path /usr/share/baresip
# audio_player alsa,default
audio_player aufile,/root/.baresip/wire3.wav
# audio_source alsa,default
audio_source aufile,/root/.baresip/wire2.wav
#audio_alert alsa,default
audio_alert null
#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_buffer_mode fixed # fixed, adaptive
audio_silence -35.0 # in [dB]
audio_telev_pt 101 # payload type for telephone-event
# Video
#video_source v4l2,/dev/video0
#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_ports 12000-12500 # 只使用 501 个端口
#rtp_bandwidth 512-1024 # [kbit/s]
audio_jitter_buffer_type fixed # off, fixed, adaptive
audio_jitter_buffer_ms 100-200 # Min. - Max. [ms]
audio_jitter_buffer_size 50 # [packets]
video_jitter_buffer_type fixed # off, fixed, adaptive
video_jitter_buffer_ms 100-200 # Min. - Max. [ms]
video_jitter_buffer_size 250 # [packets]
rtp_stats no
#rtp_timeout 60
#avt_bundle no
#rtp_rxmode main
# Network
#dns_server 1.1.1.1:53
#dns_server 1.0.0.1:53
#dns_fallback 8.8.8.8:53
#dns_getaddrinfo no
#net_interface eth0
# Play tones
#file_ausrc aufile
#file_srate 16000
#file_channels 1
#------------------------------------------------------------------------------
# Modules
module_path /usr/lib/aarch64-linux-gnu/baresip/modules
# UI Modules
module stdio.so
#module cons.so
#module evdev.so
#module httpd.so
# Audio codec Modules (in order)
#module opus.so
#module amr.so
#module g7221.so
#module g722.so
#module g726.so
module g711.so
#module l16.so
#module codec2.so
# Audio filter Modules (in encoding order)
module auconv.so
module auresamp.so
#module vumeter.so
#module sndfile.so
#module plc.so
#module webrtc_aec.so
# Audio driver Modules
module alsa.so
#module pulse.so
#module pipewire.so
#module jack.so
#module portaudio.so
#module aubridge.so
module aufile.so
#module ausine.so
# Video codec Modules (in order)
#module avcodec.so
#module vp8.so
#module vp9.so
# Video filter Modules (in encoding order)
#module selfview.so
#module snapshot.so
#module swscale.so
#module vidinfo.so
#module avfilter.so
# Video source modules
#module v4l2.so
#module vidbridge.so
# Video display modules
#module directfb.so
#module x11.so
#module sdl.so
#module fakevideo.so
# Audio/Video source modules
#module avformat.so
#module gst.so
# Compatibility modules
#module ebuacip.so
module uuid.so
# Media NAT modules
module stun.so
module turn.so
module ice.so
#module natpmp.so
#module pcp.so
# Media encryption modules
#module srtp.so
#module dtls_srtp.so
#module gzrtp.so
#------------------------------------------------------------------------------
# Application Modules
module_app account.so
module_app contact.so
module_app debug_cmd.so
#module_app echo.so
#module_app gtk.so
module_app menu.so
#module_app mwi.so
#module_app presence.so
#module_app serreg.so
#module_app syslog.so
#module_app mqtt.so
#module_app ctrl_tcp.so
#module_app ctrl_dbus.so
#module_app httpreq.so
module_app netroam.so
#------------------------------------------------------------------------------
# Module parameters
# DTLS SRTP parameters
#dtls_srtp_use_ec prime256v1
# UI Modules parameters
cons_listen 0.0.0.0:5555 # cons - Console UI UDP/TCP sockets
http_listen 0.0.0.0:8000 # httpd - HTTP Server
ctrl_tcp_listen 0.0.0.0:4444 # ctrl_tcp - TCP interface JSON
evdev_device /dev/input/event0
# Opus codec parameters
opus_bitrate 28000 # 6000-510000
#opus_stereo yes
#opus_sprop_stereo yes
#opus_cbr no
#opus_inbandfec no
#opus_dtx no
#opus_mirror no
#opus_complexity 10
#opus_application audio # {voip,audio}
#opus_samplerate 48000
#opus_packet_loss 10 # 0-100 percent (expected packet loss)
# Opus Multistream codec parameters
#opus_ms_channels 2 #total channels (2 or 4)
#opus_ms_streams 2 #number of streams
#opus_ms_c_streams 2 #number of coupled streams
vumeter_stderr yes
#jack_connect_ports yes
# Selfview
video_selfview window # {window,pip}
#selfview_size 64x64
# Menu
#redial_attempts 0 # Num or <inf>
#redial_delay 5 # Delay in seconds
#ringback_disabled no
#statmode_default off
#menu_clean_number no
#sip_autoanswer_method rfc5373 # {rfc5373,call-info,alert-info}
#ring_aufile ring.wav
#hangup_aufile none
#callwaiting_aufile callwaiting.wav
#ringback_aufile ringback.wav
#notfound_aufile notfound.wav
#busy_aufile busy.wav
#error_aufile error.wav
#sip_autoanswer_aufile autoanswer.wav
#menu_max_earlyaudio 32
#menu_max_earlyvideo_rx 32
#menu_max_earlyvideo_tx 32
#menu_message_tone yes
# GTK
#gtk_clean_number no
#gtk_use_status_icon yes
gtk_use_window yes
# avcodec
#avcodec_h264enc libx264
#avcodec_h264dec h264
#avcodec_h265enc libx265
#avcodec_h265dec hevc
#avcodec_hwaccel vaapi
#avcodec_profile_level_id 42002a
#avcodec_keyint 10
# vp8
#vp8_enc_threads 1
#vp8_enc_cpuused 16 # range -16..16, greater 0 increases speed over quality
# ctrl_dbus
#ctrl_dbus_use system # system, session
# mqtt
#mqtt_broker_host sollentuna.example.com
#mqtt_broker_port 1883
#mqtt_broker_cafile /path/to/broker-ca.crt # set this to enforce TLS
#mqtt_broker_clientid baresip01 # has to be unique
#mqtt_broker_user user
#mqtt_broker_password pass
#mqtt_basetopic baresip/01
# sndfile
#snd_path /tmp
# EBU ACIP
#ebuacip_jb_type fixed # auto,fixed
# HTTP request module
#httpreq_ca trusted1.pem
#httpreq_ca trusted2.pem
#httpreq_dns 1.1.1.1
#httpreq_dns 8.8.8.8
#httpreq_hostname myserver
#httpreq_cert cert.pem
#httpreq_key key.pem
# avformat
#avformat_hwaccel vaapi
#avformat_inputformat mjpeg
#avformat_decoder mjpeg
#avformat_pass_through yes
#avformat_rtsp_transport udp
# ice
#ice_policy all # all, relay (candidates)
Below is the command I executed:
podman run --network host -v /path/to/baresip:/root/.baresip --rm -it baresip /bin/bash
baresip -f .baresip -e "/dial sip:1002@100.113.229.80"
Below is the log content.
root@localhost:~# baresip -f .baresip -e "/dial sip:1002@100.113.229.80"
baresip v4.2.0 Copyright (C) 2010 - 2025 Alfred E. Heggestad et al.
Local network addresses:
enp0s1: 192.168.127.2
enp0s1: fe80::2a87:800:91c8:6ee4
ua: adding SIP CA file: /etc/ssl/certs/ca-certificates.crt
ua: adding SIP CA path: /etc/ssl/certs
aucodec: PCMU/8000/1
aucodec: PCMA/8000/1
aufilt: auconv
aufilt: auresamp
ausrc: alsa
auplay: alsa
ausrc: aufile
auplay: aufile
medianat: stun
medianat: turn
medianat: ice
Populated 1 account
Populated 3 contacts
Populated 2 audio codecs
Populated 2 audio filters
Populated 0 video codecs
Populated 0 video filters
baresip is ready.
/dial sip:1002@100.113.229.80
1001@100.113.229.80: fallback selection
call uri: sip:1002@100.113.229.80
call: connecting to 'sip:1002@100.113.229.80'..
call id: 9b68eb4e4636be43
1001@100.113.229.80: (prio 0) {0/UDP/v4} 200 OK (Asterisk PBX 22.6.0) [1 binding]
All 1 useragent registered successfully! (218 ms)
call: SIP Progress: 180 Ringing (/)
aufile: writing speaker audio to /root/.baresip/wire3.wav
1001@100.113.229.80: Call answered: sip:1002@100.113.229.80
stream: update 'audio'
audio_recv: Set audio decoder: PCMU 8000Hz 1ch
aufile: writing speaker audio to /root/.baresip/wire3.wav
audio_recv: player started with sample format S16LE
audio: Set audio encoder: PCMU 8000Hz 1ch
aufile: loading input file '/root/.baresip/wire2.wav'
aufile: /root/.baresip/wire2.wav: 8000 Hz, 1 channels, S16LE
aufile: audio ptime=20 sampc=160
aufile: read end of file
aufile: loaded 498880 bytes
audio: source started with sample format S16LE
audio tx pipeline: aufile ---> aubuf ---> auconv ---> auresamp ---> PCMU
audio rx pipeline: aufile <--- aubuf <--- auconv <--- auresamp <--- PCMU
1001@100.113.229.80: Call established: sip:1002@100.113.229.80
terminated by signal 20 (bit/s)
ua: stop all (forced=0)
sip:1001@100.113.229.80: Call with sip:1002@100.113.229.80 terminated (duration: 5 secs)