NEW: telephony/pjsua

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NEW: telephony/pjsua

Deanna Phillips-2
This is a command line SIP user agent.  

I've only tested it on i386, more testing appreciated.  If you
experience sound problems, try the stereo FLAVOR or using the
--clock-rate option.

There's a sample config for Free World Dialup in pkg/MESSAGE.

Docs: http://www.pjsip.org/pjsua.htm


pjsua.tar.gz (13K) Download Attachment
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Re: NEW: telephony/pjsua

Claudio Correa
Hi Deanna,

        I tested pjsua for amd64 with asterisk in the local net, also
with stereo FLAVOR. Build, install/deinstall ok, but has problems in the
reproduction of the received sound. There are some delay and is
fragmented completely.

regards
OpenBSD 4.0-current (GENERIC) #824: Tue Feb  6 18:40:57 MST 2007
[hidden email]:/usr/src/sys/arch/amd64/compile/GENERIC
cpu0: AMD Athlon(tm) 64 Processor 3200+, 798.04 MHz

auich0 at pci0 dev 6 function 0 "NVIDIA nForce3 AC97" rev 0xa2: irq 11,
nForce3 AC97 ac97: codec id 0x41445374 (Analog Devices AD1981B)
ac97: codec features headphone, 20 bit DAC, No 3D Stereo


> This is a command line SIP user agent.  
>
> I've only tested it on i386, more testing appreciated.  If you
> experience sound problems, try the stereo FLAVOR or using the
> --clock-rate option.
>
> There's a sample config for Free World Dialup in pkg/MESSAGE.
>
> Docs: http://www.pjsip.org/pjsua.htm


--
\\``""""''// .....................................................
  | o` _o |  Prof. Claudio Correa - Computer Science - 10 Years
  \   v   /  Pontifical Catholic University of Minas Gerais
   \  U  /  PUC Minas - The best Private University of Brazil
     -v-    Campus at Po_os de Caldas - Po_os de Caldas - MG / Brazil
Av. Padre Francis Cletus Cox, 1661 - CEP: 37701-355
Phone: +55 (0)35 3729-9269 Fax: +55 (0)35 3729-9201
mailto:[hidden email]     http://www.pucpcaldas.br
________________________________________________________________________

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Re: NEW: telephony/pjsua

Stuart Henderson
On 2007/06/06 10:45, Claudio Correa wrote:
>
> I tested pjsua for amd64 with asterisk in the local net, also
> with stereo FLAVOR. Build, install/deinstall ok, but has problems in the
> reproduction of the received sound. There are some delay and is
> fragmented completely.

I see (er, hear) the same thing on i386 and amd64, have tried calls
pjsua<>pjsua, pjsua->asterisk->snom, pjsua->asterisk IVR, --clock-rate
48000 to match output devices, tried stereo/not.

Audio transmitted from pjsua is ok, audio recorded by pjsua to wav file
and replayed with mplayer is ok, audio output from pjsua to /dev/audio is
terribly chopped-up (silence most of the time with maybe 4 short bursts
a second of audio; not good enough to hear anything). As soon as the
app starts up on my amd64 box I get short bursts of clicks at about
the same interval as audio is played (4x/sec or so - the interval
varies depending on clock-rate).

Any clues? It uses PortAudio if that rings any bells with anyone...

i386 (X40):

OpenBSD 4.1-current (GENERIC) #211: Sat Jun  2 15:53:04 MDT 2007
    [hidden email]:/usr/src/sys/arch/i386/compile/GENERIC
cpu0: Intel(R) Pentium(R) M processor 1200MHz ("GenuineIntel" 686-class) 1.20 GHz
cpu0: FPU,V86,DE,PSE,TSC,MSR,MCE,CX8,APIC,SEP,MTRR,PGE,MCA,CMOV,PAT,CFLUSH,DS,ACPI,MMX,FXSR,SSE,SSE2,TM,SBF,EST,TM2
avail mem = 499945472 (476MB)
mainbus0 at root
bios0 at mainbus0: AT/286+ BIOS, date 03/01/06, BIOS32 rev. 0 @ 0xfd740, SMBIOS rev. 2.33 @ 0xe0010 (56 entries)
bios0: IBM 23718EG
[...]
auich0 at pci0 dev 31 function 5 "Intel 82801DB AC97" rev 0x01: irq 11, ICH4 AC97
ac97: codec id 0x41445374 (Analog Devices AD1981B)
ac97: codec features headphone, 20 bit DAC, No 3D Stereo
audio0 at auich0

amd64:

OpenBSD 4.1-current (GENERIC) #1053: Sat Jun  2 12:00:55 MDT 2007
    [hidden email]:/usr/src/sys/arch/amd64/compile/GENERIC
real mem = 1072164864 (1022MB)
avail mem = 1024196608 (976MB)
mainbus0 at root
bios0 at mainbus0: SMBIOS rev. 2.2 @ 0xf0000 (42 entries)
acpi0 at mainbus0: rev 0
acpi0: tables DSDT FACP SSDT MCFG APIC
acpitimer at acpi0 not configured
acpiprt0 at acpi0: bus 0 (PCI0)
acpiprt1 at acpi0: bus 1 (HUB0)
acpicpu at acpi0 not configured
acpicpu at acpi0 not configured
acpitz at acpi0 not configured
acpibtn at acpi0 not configured
cpu0 at mainbus0: (uniprocessor)
cpu0: AMD Athlon(tm) 64 Processor 3700+, 2211.59 MHz
cpu0: FPU,VME,DE,PSE,TSC,MSR,PAE,MCE,CX8,APIC,SEP,MTRR,PGE,MCA,CMOV,PAT,PSE36,CFLUSH,MMX,FXSR,SSE,SSE2,SSE3,NXE,MMXX,FFXSR,LONG,3DNOW2,3DNOW
cpu0: 64KB 64b/line 2-way I-cache, 64KB 64b/line 2-way D-cache, 1MB 64b/line 16-way L2 cache
cpu0: ITLB 32 4KB entries fully associative, 8 4MB entries fully associative
cpu0: DTLB 32 4KB entries fully associative, 8 4MB entries fully associative
cpu0: AMD erratum 89 present, BIOS upgrade may be required
[...]
auich0 at pci0 dev 4 function 0 "NVIDIA nForce4 AC97" rev 0xa2: irq 10, nForce4 AC97
ac97: codec id 0x414c4760 (Avance Logic ALC655 rev 0)
audio0 at auich0

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Re: NEW: telephony/pjsua

Deanna Phillips-2
If you are experiencing sound problems, please do the following:

Run the sound test in the build directory.  It will be some
place like this, depending on your platform:

${WRKSRC}/pjsip-apps/bin/samples/sndtest-i386-unknown-openbsd4.1

Then paste me the output.

Also, make sure that you hang up ( 'h' ) all calls before
quitting.  Dangling calls will cause audio breakups or complete
silence.

More sound troubleshooting tips here:

http://www.pjsip.org/trac/wiki/audio-problem-breakups

Thanks!

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Re: NEW: telephony/pjsua

Claudio Correa
Hi Deanna,

        Result of tests

./pjsua --play-file /tmp/pcm.wav
....
You have 0 active call
>>> cl
Conference ports:
Port #00[16KHz/20ms]         Master/sound  transmitting to:
Port #01[16KHz/20ms]         splitcomb-ch  transmitting to:
Port #02[ 8KHz/20ms]         /tmp/pcm.wav  transmitting to:

>>> cc 1 0
16:00:49.409  conference.c Port 1 (splitcomb-ch) transmitting to port
0 (Master/sound)
Success

But, I couldn't hear the file played to the speaker.

Then, with
./playfile /tmp/pcm.wav

ok, the file played properly to the speaker

From: http://www.pjsip.org/trac/wiki/audio-check-play
"The difference between pjsua and playfile program is the lack of
conference bridge in playfile." ????

At  ${WRKSRC}/pjsip-apps/bin/samples
./sndtest-x86_64-unknown-openbsd4.0 >> /tmp/sndtest.log
sndtest.log attached

Regards


On Fri, 08 Jun 2007 10:43:22 +0000
Deanna Phillips <[hidden email]> wrote:

> If you are experiencing sound problems, please do the following:
>
> Run the sound test in the build directory.  It will be some
> place like this, depending on your platform:
>
> ${WRKSRC}/pjsip-apps/bin/samples/sndtest-i386-unknown-openbsd4.1
>
> Then paste me the output.
>
> Also, make sure that you hang up ( 'h' ) all calls before
> quitting.  Dangling calls will cause audio breakups or complete
> silence.
>
> More sound troubleshooting tips here:
> http://www.pjsip.org/trac/wiki/audio-problem-breakups
>
> Thanks!
>

sndtest.log (29K) Download Attachment