Re: NEW: telephony/pjsua
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 +
Deanna Phillips [EMAIL PROTECTED] 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
Description: Binary data
Re: NEW: telephony/pjsua
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 [EMAIL PROTECTED]:/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:[EMAIL PROTECTED] http://www.pucpcaldas.br
Re: NEW: telephony/pjsua
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 pjsuapjsua, 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 [EMAIL PROTECTED]:/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 [EMAIL PROTECTED]:/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 @ 0xf (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
NEW: telephony/pjsua
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 Description: Binary data
