Linphone and PulseAudio only static and clicks instead of voice
Peter Gregory
My USB audio card only seems to support 44khz for recording and playback.
I'm running linphonec 3.5.2 with Debian Linux beaglebone 3.8.13-bone72 #1 SMP Tue Jun 16 21:36:04 UTC 2015 armv7l GNU/Linux
My goal is to have peer-to-peer SIP communications between my Mac running Linphone and my BBB running Linphone.
They are able to communicate and connect to each other.
The audio on both sides comes across and clicks and static noises.
Any help would be appreciated.
The negotiation for audio looks like this:
ortp-message-cb_snd123456xx (id=1)
Connected.
linphonec> ortp-message-Call 0x6b8a8: moving from state LinphoneCallIncomingReceived to LinphoneCallConnected
Call 1 with <sip:pgre...@10.128.84.155> connected.
ortp-message-Audio bandwidth for this call is 44
ortp-message-speex_lib_ctl does not support SPEEX_LIB_CPU_FEATURE_NEON
ortp-message-cb_nict_kill_transaction (id=1)
ortp-message-eXosip: timer sec:0 usec:648457!
ortp-message-Payload's bitrate is 44000
ortp-message-Setting audio encoder network bitrate to 44000
ortp-message-ms_filter_link: MSAlsaRead:0x70e98,0-->MSSpeexEC:0x81250,1
ortp-message-ms_filter_link: MSSpeexEC:0x81250,1-->MSVolume:0x6fa98,0
ortp-message-ms_filter_link: MSVolume:0x6fa98,0-->MSSpeexEnc:0x6bcd0,0
ortp-message-ms_filter_link: MSSpeexEnc:0x6bcd0,0-->MSRtpSend:0x70f20,0
ortp-message-ms_filter_link: MSRtpRecv:0x789b8,0-->MSSpeexDec:0x6fa20,0
ortp-message-ms_filter_link: MSSpeexDec:0x6fa20,0-->MSDtmfGen:0x810b8,0
ortp-message-ms_filter_link: MSDtmfGen:0x810b8,0-->MSVolume:0x6be68,0
ortp-message-ms_filter_link: MSVolume:0x6be68,0-->MSEqualizer:0x6bed0,0
ortp-message-ms_filter_link: MSEqualizer:0x6bed0,0-->MSSpeexEC:0x81250,0
ortp-message-ms_filter_link: MSSpeexEC:0x81250,0-->MSAlsaWrite:0x76540,0
ortp-message-Initializing speex echo canceler with framesize=64, filterlength=4000, delay_samples=0
ortp-message-Setting maxbitrate=28000 to speex encoder.
ortp-message-Using bitrate 27800 for speex encoder, ip bitrate is 43600
ortp-message-Priority used: 99
ortp-message-Audio MSTicker setpriority() failed: Permission denied, nevermind.
ortp-message-Filter MSRtpRecv is already being scheduled; nothing to do.
ortp-message-Call 0x6b8a8: moving from state LinphoneCallConnected to LinphoneCallStreamsRunning
Media streams established with <sip:pgre...@10.128.84.155> for call 1.
ortp-message-call answered.
linphonec> ortp-message-alsa_open_r: opening default at 16000Hz, bits=16, stereo=0
xcb_connection_has_error() returned true
ortp-message-Received message:
ACK sip:deb...@10.128.84.154 SIP/2.0
Via: SIP/2.0/UDP 10.128.84.155:5060;rport;branch=z9hG4bK.FHinVVo8u
From: <sip:pgre...@10.128.84.155>;tag=gm8PDvkzI
To: <sip:deb...@10.128.84.154>;tag=1705673520
CSeq: 20 ACK
Call-ID: 7x78li-cQI
Max-Forwards: 70
ortp-message-Message received from: 10.128.84.155:5060
ortp-message-Message received from: 10.128.84.155:5060
ortp-message-MESSAGE REC. CALLID:7x78li-cQI
ortp-message-Message received from: 10.128.84.155:5060
ortp-message-This is a request
ortp-message-linphone process event get a message 15
ortp-message-CALL_ACK
ortp-message-eXosip: timer sec:0 usec:547862!
ortp-warning-alsa_set_params: periodsize:512 Using 512
ortp-warning-alsa_set_params: period:8 Using 8
ortp-message-alsa_open_r: Audio params set
ortp-message-ms_ticker_set_time_func: ticker updated.
ortp-warning-Getting reference signal but no echo to synchronize on.
ortp-warning-Not enough ref samples, using zeroes
ortp-message-alsa_open_w: opening default at 16000Hz, bits=16, stereo=0
xcb_connection_has_error() returned true
ortp-warning-alsa_set_params: periodsize:512 Using 512
ortp-warning-alsa_set_params: period:8 Using 8
ortp-message-bandwidth usage: audio=[d=0.0,u=0.0] video=[d=0.0,u=0.0] kbit/sec
ortp-message-Thread processing load: audio=129.954559 video=0.000000
ortp-message-alsa_open_w: Audio params set
My configuration for Linphone is the default it creates when started:
debian@beaglebone:~$ cat .linphonerc
[sip]
media_encryption=none
guess_hostname=1
contact=sip:debian@unknown-host
inc_timeout=15
use_info=0
use_rfc2833=0
use_ipv6=0
register_only_when_network_is_up=1
[net]
download_bw=0
upload_bw=0
firewall_policy=0
mtu=0
[call_log_0]
dir=1
status=0
from=<sip:pgre...@10.128.84.155>
to=<sip:deb...@10.128.84.154>
start_date=Thu Oct 29 12:23:34 2015
duration=14
quality=1.889588
[rtp]
audio_rtp_port=7078
video_rtp_port=9078
audio_jitt_comp=60
video_jitt_comp=60
nortp_timeout=30
[sound]
remote_ring=/usr/share/sounds/linphone/ringback.wav
[video]
size=cif
display=0
capture=0
show_local=0
self_view=1
[audio_codec_0]
mime=speex
rate=32000
enabled=1
[audio_codec_1]
mime=speex
rate=16000
enabled=1
[audio_codec_2]
mime=speex
rate=8000
enabled=1
[audio_codec_3]
mime=GSM
rate=8000
enabled=1
[audio_codec_4]
mime=PCMU
rate=8000
enabled=1
[audio_codec_5]
mime=PCMA
rate=8000
enabled=1
[audio_codec_6]
mime=L16
rate=44100
enabled=0
[audio_codec_7]
mime=L16
rate=44100
enabled=0
[audio_codec_8]
mime=G722
rate=8000
enabled=0
[video_codec_0]
mime=MP4V-ES
rate=90000
enabled=1
recv_fmtp=profile-level-id=3
[video_codec_1]
mime=H263-1998
rate=90000
enabled=1
recv_fmtp=CIF=1;QCIF=1
[video_codec_2]
mime=theora
rate=90000
enabled=0
[video_codec_3]
mime=x-snow
rate=90000
enabled=0
[video_codec_4]
mime=H263
rate=90000
enabled=0
[call_log_1]
dir=1
status=0
from=<sip:pgre...@10.128.84.155>
to=<sip:deb...@10.128.84.154>
start_date=Thu Oct 29 12:21:51 2015
duration=29
quality=3.969424
Peter Gregory
I modified my application to just use the ALSA API and did not install the pulse audio drivers.
Just using ALSA works fine with linphonec.
Recording and playback work as expected.