On Thu, Jan 03, 2013 at 08:15:41AM +0100, Matthias Apitz wrote:
>
> Hi,
>
> El d?a Wednesday, January 02, 2013 a las 08:19:11PM -0800, Joseph Olatt
> escribi?:
>
> >
> > I've been trying to get baresip to work on my FreeBSD 9.x laptop and
> > haven't had much success. I register successfully to callcentric.com and
> > it appears that I can connect and there is a stream of data coming
> > through based on the status display:
> >
> > [0:00:08] audio=0/0 (bit/s) [ ]
>
> The display shows zero audio data!
>
> > However, there is no sound.
>
> Have you tried the local audio loop with pressing the single letter 'a'?
>
> > Is anybody on the list successfully using baresip? If so, could they
> > please provide some pointers on how to get sound?
>
> I'm attaching my config file which works fine; in your config file it
> looks stange to me:
>
> > # Audio
> > audio_dev /dev/audio0.0
>
> do you have such a device file '/dev/audio0.0'?
>
> > # Audio codec Modules (in order)
> > #module g7221.so
> > #module g722.so
> > module g711.so
> > #module gsm.so
> > #module l16.so
> > #module speex.so
> > #module celt.so
> > #module bv32.so
> >
> > # Audio filter Modules (in order)
> > # NOTE: AEC should be before Preproc
> > #module sndfile.so
> > #module speex_aec.so
> > #module speex_pp.so
> > #module speex_resamp.so
> > #module plc.so
> >
> > # Audio driver Modules
> > #module oss.so
> > #module alsa.so
> > #module portaudio.so
> > #module gst.so
>
> you have no audio driver loaded, try 'oss.so'
>
> Once you get the local loop working you could contact me off-list for my
> SIP and try to call me.
>
> HIH
>
> matthias
> --
> Sent from my FreeBSD netbook
>
> Matthias Apitz | - No system with backdoors like Apple/Android
> E-mail: g...@unixarea.de | - No HTML/RTF in E-mail
> WWW: http://www.unixarea.de/ | - No proprietary attachments
> phone: +49-170-4527211 | - Respect for open standards
> #
> # baresip configuration
> #
>
> #------------------------------------------------------------------------------
>
> # Core
> poll_method poll # poll, select, epoll ..
>
> # Input
> input_device /dev/event0
> input_port 5555
>
> # SIP
> sip_trans_bsize 128
> #sip_listen 127.0.0.1:5050
>
> # Audio
> audio_dev /dev/dsp
> audio_srate 8000-48000
> audio_channels 1-2
> #audio_aec_length 128 # [ms]
>
> # Video
> video_dev /dev/video0
> video_size 352x288
> video_bitrate 384000
> video_fps 25
> #video_selfview window # {window,pip}
>
> # AVT - Audio/Video Transport
> rtp_tos 184
> #rtp_ports 10000-20000
> rtp_ports 1024-1030
> #rtp_bandwidth 512-1024 # [kbit/s]
> rtcp_enable yes
> rtcp_mux no
> jitter_buffer_delay 5-10 # frames
>
> # Network
> #dns_server 10.0.0.1:53
>
> #------------------------------------------------------------------------------
> # Modules
>
> module_path /usr/local/lib/baresip/modules
>
> # UI Modules
> module stdio.so
> module cons.so
> #module evdev.so
>
> # Audio codec Modules (in order)
> #module g7221.so
> #module g722.so
> module g711.so
> #module gsm.so
> #module l16.so
> #module speex.so
> #module celt.so
> #module bv32.so
>
> # Audio filter Modules (in order)
> # NOTE: AEC should be before Preproc
> #module sndfile.so
> #module speex_aec.so
> #module speex_pp.so
> #module speex_resamp.so
> #module plc.so
>
> # Audio driver Modules
> module oss.so
> #module alsa.so
> #module portaudio.so
> #module gst.so
>
> # Video codec Modules (in order)
> module avcodec.so
> #module vpx.so
>
> # Video source modules
> module v4l2.so
> #module avformat.so
> #module v4l.so
>
> # Video display modules
> module x11.so
> # module sdl.so
>
> # Media NAT modules
> module stun.so
> module turn.so
> module ice.so
>
> # Media encoding modules
> #module srtp.so
>
> # Other modules
> #module natbd.so
>
> #------------------------------------------------------------------------------
> # Module parameters
>
>
> # Speex codec parameters
> speex_quality 7 # 0-10
> speex_complexity 7 # 0-10
> speex_enhancement 0 # 0-1
> speex_vbr 0 # Variable Bit Rate 0-1
> speex_vad 0 # Voice Activity Detection 0-1
> speex_agc_level 8000
>
> # NAT Behavior Discovery
> natbd_server creytiv.com
> natbd_interval 600 # in seconds
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Hi Matthias and Hugo,
Thank you for your responses. After reading them I realized that I need
to check one of the config options for the audio driver. For the benefit
of future users of baresip on FreeBSD:
cd /usr/ports/audio/baresip
> make showconfig
===> The following configuration options are available for
baresip-0.4.0:
CELT=off: CELT audio codec
CONS=on: Console input driver
DOCS=on: Build and/or install documentation
FFMPEG=off: FFmpeg (WMA, AIFF, AC3, APE...)
G711=on: G.711 audio codec
G722=on: G.722 audio codec
GSM=off: GSM codec support
GSTREAMER=off: Multimedia via GStreamer
ILBC=off: iLBC audio codec
L16=on: L16 audio codec
OPUS=off: Opus audio codec
OSS=on: Open Sound System <<-- Enable
PORTAUDIO=on: PortAudio library <<-- Enable
SDL=off: Simple Direct Media Layer
SNDFILE=off: libsndfile support
SPEEX=off: Speex audio format
SRTP=off: Secure RTP module
STDIO=on: stdio input driver
UUID=off: UUID module
V4L=off: Video4Linux module
V4L2=off: Video4Linux2 module
X11=off: X11 (graphics) support
===> Use 'make config' to modify these settings
Needed to enable OSS or PORTAUDIO. I enabled both and tested by
activating each one at a time in ~/.baresip/config file. Both
worked.
I'm still not sure if speaking into the microphone is working. As soon
as I find someone to communicate with using SIP, I'll find out.
Thanks to both of you for your assistance.
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