no sound output, while dialing via mqtt command on embedded arm device #3478
Description
Hi,
i'm using baresip within a embedded device. Baresip is running as an application, my frontend application interfaces baresip via MQTT. Both applications are executed on the embedded device. I'm using a mosquitto, which is also executed on the device.
While porting to a new platform (arm device, based on freescale imx6), i'm observing problems. My application sends a dial command and baresip starts the call. But while dialing process, i don't get any dial tone. If i accept the call, there is also no sound (no mic sound, no speaker sound).
If i send the same MQTT command via mosquitto_pub command from, baresip is working fine with sound output. I can also start calls from the baresip CLI frontend, or executing the application on my host (while connecting the broker on the device). For the tests, the same baresip instance is used/baresip was never restarted. If i switch back to the application on the device, the problem is still there. So i assume, baresip is not in some strange state and the problem must be related to the specific MQTT message from the application.
This setup is stable and can be reproduced every time on the device. I already analyses the MQTT packages via tcpdump, but don't see any difference. See pictures.
On the same platform i also observed a problem, while running baresip within a Boost::Process::Child. While performing the call, baresip exited with error code 7, caused by SIGBUS signal.
Unfortunately i cannot reproduce this on my host device. Does somebody have an idea, what could be the reason for this behaviour?
I'm using the current v4.0.0 release and also on an older release.
Thank you very much in advance and best regards,
Marcel
Working Frame:
Broken Frame:
Baresip log with working sound:
mqtt: got message '{"command" : "dial", "params" : "192.168.2.47", "token" : "dial" }' for topic '/baresip/01/command/c'
mqtt: got message for '/baresip/01/command/c' topic
mqtt: handle_command: cmd='dial', token='dial'
@127.0.0.1: selected for request
call: alloc with params laddr=192.168.2.41, af=AF_INET, use_rtp=1
call: use_video=0
call: connecting to 'sip:192.168.2.47'..
call: SIP Progress: 180 Ringing (/)
stream: disable audio RTP sender
stream: disable audio RTP receiver
alsa: reset: srate=8000, ch=1, num_frames=320, pcmfmt=S16_LE
alsa: playback started (sysdefault:CARD=imx6qapalissgtl)
call: SIP Progress: 180 Ringing (/)
stream: disable audio RTP sender
stream: disable audio RTP receiver
call: SIP Progress: 180 Ringing (/)
stream: disable audio RTP sender
stream: disable audio RTP receiver
call: SIP Progress: 180 Ringing (/)
stream: disable audio RTP sender
stream: disable audio RTP receiver
call: got SDP answer (244 bytes)
@127.0.0.1: Call answered: sip:192.168.2.47
alsa: stopping playback thread (sysdefault:CARD=imx6qapalissgtl)
call: update media
stream: update 'audio'
stream: disable audio RTP receiver
stream: disable audio RTP sender
stream: enable audio RTP receiver
stream: enable audio RTP sender
stream: audio: starting RTCP with remote 192.168.2.47:56597
audio: update
audio_recv: Set audio decoder: PCMU 8000Hz 1ch
alsa: reset: srate=8000, ch=1, num_frames=160, pcmfmt=S16_LE
alsa: playback started (sysdefault:CARD=imx6qapalissgtl)
audio_recv: player started with sample format S16LE
audio: Set audio encoder: PCMU 8000Hz 1ch
stream: enable audio RTP receiver
alsa: reset: srate=8000, ch=1, num_frames=160, pcmfmt=S16_LE
alsa: recording started (sysdefault:CARD=Device) format=S16LE
audio: source started with sample format S16LE
stream: enable audio RTP sender
audio tx pipeline: alsa ---> aubuf ---> auconv ---> auresamp ---> mixausrc ---> PCMU
audio rx pipeline: alsa <--- aubuf <--- auconv <--- auresamp <--- mixausrc <--- PCMU
call: timer started
@127.0.0.1: Call established: sip:192.168.2.47
auresamp: resample encoder 8000/1 --> 8000/1
sip:192.168.2.47: session closed: Connection reset by peer [104]
stream: disable audio RTP sender
alsa: stopping recording thread (sysdefault:CARD=Device)
stream: disable audio RTP receiver
alsa: stopping playback thread (sysdefault:CARD=imx6qapalissgtl)
sip:127.0.0.1: Call with sip:192.168.2.47 terminated (duration: 1 sec)
stream: disable audio RTP sender
stream: disable audio RTP receiver
audio: destroyed (started=0)
stream: disable audio RTP sender
stream: disable audio RTP receiver
Baresip log with broken sound:
mqtt: got message '{"command" : "dial", "params" : "192.168.2.47", "token" : "dial" }' for topic '/baresip/01/command/c'
mqtt: got message for '/baresip/01/command/c' topic
mqtt: handle_command: cmd='dial', token='dial'
@127.0.0.1: selected for request
call: alloc with params laddr=192.168.2.41, af=AF_INET, use_rtp=1
call: use_video=0
call: connecting to 'sip:192.168.2.47'..
call: SIP Progress: 180 Ringing (/)
stream: disable audio RTP sender
stream: disable audio RTP receiver
alsa: reset: srate=8000, ch=1, num_frames=320, pcmfmt=S16_LE
alsa: playback started (sysdefault:CARD=imx6qapalissgtl)
call: SIP Progress: 180 Ringing (/)
stream: disable audio RTP sender
stream: disable audio RTP receiver
call: SIP Progress: 180 Ringing (/)
stream: disable audio RTP sender
stream: disable audio RTP receiver
call: SIP Progress: 180 Ringing (/)
stream: disable audio RTP sender
stream: disable audio RTP receiver
call: got SDP answer (244 bytes)
@127.0.0.1: Call answered: sip:192.168.2.47
alsa: stopping playback thread (sysdefault:CARD=imx6qapalissgtl)
call: update media
stream: update 'audio'
stream: disable audio RTP receiver
stream: disable audio RTP sender
stream: enable audio RTP receiver
stream: enable audio RTP sender
stream: audio: starting RTCP with remote 192.168.2.47:57131
audio: update
audio_recv: Set audio decoder: PCMU 8000Hz 1ch
alsa: reset: srate=8000, ch=1, num_frames=160, pcmfmt=S16_LE
alsa: playback started (sysdefault:CARD=imx6qapalissgtl)
audio_recv: player started with sample format S16LE
audio: Set audio encoder: PCMU 8000Hz 1ch
stream: enable audio RTP receiver
alsa: reset: srate=8000, ch=1, num_frames=160, pcmfmt=S16_LE
alsa: recording started (sysdefault:CARD=Device) format=S16LE
audio: source started with sample format S16LE
stream: enable audio RTP sender
audio tx pipeline: alsa ---> aubuf ---> auconv ---> auresamp ---> mixausrc ---> PCMU
audio rx pipeline: alsa <--- aubuf <--- auconv <--- auresamp <--- mixausrc <--- PCMU
call: timer started
@127.0.0.1: Call established: sip:192.168.2.47
auresamp: resample encoder 8000/1 --> 8000/1
rtprecv: incoming rtp for 'audio' established, receiving from 192.168.2.47:57130
sip:192.168.2.47: session closed: Connection reset by peer [104]
stream: disable audio RTP sender
alsa: stopping recording thread (sysdefault:CARD=Device)
stream: disable audio RTP receiver
alsa: stopping playback thread (sysdefault:CARD=imx6qapalissgtl)
stream: disable audio RTP sender
stream: disable audio RTP receiver
audio: destroyed (started=0)
stream: disable audio RTP sender
stream: disable audio RTP receiver
mqtt: got message '{
"command" : "hangup",
"params" : "",
"token" : "hangup"
}' for topic '/baresip/01/command/c'
mqtt: got message for '/baresip/01/command/c' topic
mqtt: handle_command: cmd='hangup', token='hangup'
Used config file:
#
# baresip configuration
#
#------------------------------------------------------------------------------
# SIP
sip_listen 0.0.0.0:5060
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/local/share/baresip
audio_player alsa,sysdefault:CARD=imx6qapalissgtl
audio_source alsa,sysdefault:CARD=Device
audio_alert alsa,sysdefault:CARD=imx6qapalissgtl
#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_bandwidth 512-1024 # [kbit/s]
audio_jitter_buffer_type fixed # off, fixed, adaptive
audio_jitter_buffer_delay 5-10 # (min. frames)-(max. packets)
video_jitter_buffer_type fixed # off, fixed, adaptive
video_jitter_buffer_delay 5-50 # (min. frames)-(max. packets)
rtp_stats no
#rtp_timeout 60
#avt_bundle no
# 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/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 mpa.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 mixausrc.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 multicast.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
#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
# GTK
#gtk_clean_number no
# avcodec
#avcodec_h264enc libx264
#avcodec_h264dec h264
#avcodec_h265enc libx265
#avcodec_h265dec hevc
#avcodec_hwaccel vaapi
#avcodec_profile_level_id 42002a
#avcodec_keyint 10
# ctrl_dbus
#ctrl_dbus_use system # system, session
# mqtt
mqtt_broker_host localhost
mqtt_broker_port 1883
#mqtt_broker_cafile /path/to/broker-ca.crt # set this to enforce TLS
mqtt_broker_clientid baresip-libelfinemergencyphone-main
#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
# multicast receivers (in priority order)- port number must be even
#multicast_call_prio 0
#multicast_ttl 1
#multicast_jbuf_type fixed # off, fixed, adaptive
#multicast_jbuf_delay 5-10 # frames
#multicast_listener 224.0.2.21:50000
#multicast_listener 224.0.2.21:50002
# 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)
webrtc_aec_extended_filter yes