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Date:      Thu, 3 Jan 2013 12:41:00 -0800
From:      Joseph Olatt <joji@eskimo.com>
To:        Matthias Apitz <guru@unixarea.de>, hugo@barafranca.com
Cc:        freebsd-questions@freebsd.org
Subject:   Re: audio/baresip on FreeBSD 9
Message-ID:  <20130103204100.GA1376@shellx.eskimo.com>
In-Reply-To: <20130103071541.GB1219@tiny.Sisis.de>
References:  <20130103041911.GA29654@shellx.eskimo.com> <20130103071541.GB1219@tiny.Sisis.de>

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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: guru@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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