I don't see anything wrong in the log, although if there are lots of Master/sound underflow errors that would indicate your processor can't keep up, seems very unlikely. Your default setup is using PJSIP's MME drivers. I think the only other Windows option is to use portaudio, which you can do by putting the following lines in your config_site.h file. #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 1 #define PJMEDIA_AUDIO_DEV_HAS_WMME 0 Are you using a headset device? If not, then when you do cc 0 0 you will get echo, which the AEC will try to suppress. Maybe that's the source of your "metallic" sound? You can also test by playing a WAV file to speaker and recording mic to a WAV file, using pjsua options. Bill On 3/13/2016 5:40 PM, Ebubekir DEM?R wrote: > Thanks for reply. > I recorded a sound file using windows tool in order to test audio > device. Sound which is heard in this file has high quality. > Then, I run pjsua app at another computer. There was no problem in > sound heard. In this case , > I think that reason of bad voice is choice of drivers. How do I change > the sound driver? > > pjsua log is below: > C:\Users\demir-bekir\Desktop\pjsua_desk_success\vs2\pjproject-2.4.5\pjsip-apps\bin>pjsua-i386-Win32-vc8-Debug.exe > 23:36:38.701 os_core_win32. !pjlib 2.4.5 for win32 initialized > 23:36:38.733 pjlib .select() I/O Queue created (009BCDA8) > 23:36:38.738 sip_endpoint.c .Module "mod-msg-print" registered > 23:36:38.742 pjsua_core.c .PJSUA state changed: NULL --> CREATED > 23:36:38.747 sip_endpoint.c .Module "mod-pjsua-log" registered > 23:36:38.751 sip_endpoint.c .Module "mod-tsx-layer" registered > 23:36:38.756 sip_endpoint.c .Module "mod-stateful-util" registered > 23:36:38.761 sip_endpoint.c .Module "mod-ua" registered > 23:36:38.765 sip_endpoint.c .Module "mod-100rel" registered > 23:36:38.770 sip_endpoint.c .Module "mod-pjsua" registered > 23:36:38.774 sip_endpoint.c .Module "mod-invite" registered > 23:36:38.811 wmme_dev.c ..WMME found 3 devices: > 23:36:38.814 wmme_dev.c .. dev_id 0: Wave mapper (in=2, out=2) > 23:36:38.819 wmme_dev.c .. dev_id 1: Microphone (Conexant > SmartAudio (in=2, out=0) > 23:36:38.824 wmme_dev.c .. dev_id 2: Speakers (Conexant > SmartAudio H (in=0, out=2) > 23:36:38.828 wmme_dev.c ..WMME initialized > 23:36:38.832 pjlib ..select() I/O Queue created (00A1A814) > 23:36:38.847 sip_endpoint.c .Module "mod-evsub" registered > 23:36:38.852 sip_endpoint.c .Module "mod-presence" registered > 23:36:38.856 sip_endpoint.c .Module "mod-mwi" registered > 23:36:38.860 sip_endpoint.c .Module "mod-refer" registered > 23:36:38.863 sip_endpoint.c .Module "mod-pjsua-pres" registered > 23:36:38.872 sip_endpoint.c .Module "mod-pjsua-im" registered > 23:36:38.876 sip_endpoint.c .Module "mod-pjsua-options" registered > 23:36:38.881 pjsua_core.c .1 SIP worker threads created > 23:36:38.885 pjsua_core.c .pjsua version 2.4.5 for > win32-6.2/i386/msvc-17.0 initialized > 23:36:38.889 pjsua_core.c .PJSUA state changed: CREATED --> INIT > 23:36:38.893 sip_endpoint.c Module "mod-default-handler" registered > 23:36:38.906 pjsua_core.c SIP UDP socket reachable at > 192.168.1.22:5060 <http://192.168.1.22:5060> > 23:36:38.911 udp009CEF50 SIP UDP transport started, published > address is 192.168.1.22:5060 <http://192.168.1.22:5060> > 23:36:38.916 pjsua_acc.c Adding account: id=<sip:192.168.1.22:5060 > <http://192.168.1.22:5060>> > 23:36:38.922 pjsua_acc.c .Account <sip:192.168.1.22:5060 > <http://192.168.1.22:5060>> added with id 0 > 23:36:38.926 pjsua_acc.c Modifying account 0 > 23:36:38.930 pjsua_acc.c Acc 0: setting online status to 1.. > 23:36:38.941 tcptp:5060 SIP TCP listener ready for incoming > connections at 192.168.1.22:5060 <http://192.168.1.22:5060> > 23:36:38.947 pjsua_acc.c Adding account: > id=<sip:192.168.1.22:5060;transport=TCP> > 23:36:38.952 pjsua_acc.c .Account > <sip:192.168.1.22:5060;transport=TCP> added with id 1 > 23:36:38.957 pjsua_acc.c Modifying account 1 > 23:36:38.961 pjsua_acc.c Acc 1: setting online status to 1.. > 23:36:38.965 pjsua_core.c PJSUA state changed: INIT --> STARTING > 23:36:38.970 sip_endpoint.c .Module "mod-unsolicited-mwi" registered > 23:36:38.975 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING > 23:36:38.979 main.c Ready: Success > >>>> > Account list: > [ 0] <sip:192.168.1.22:5060 <http://192.168.1.22:5060>>: does not > register > Online status: Online > *[ 1] <sip:192.168.1.22:5060;transport=TCP>: does not register > Online status: Online > Buddy list: > -none- > > +=============================================================================+ > | Call Commands: | Buddy, IM & Presence: | > Account: | > | | | > | > | m Make new call | +b Add new buddy .| +a Add > new accnt | > | M Make multiple calls | -b Delete buddy | -a Delete > accnt. | > | a Answer call | i Send IM | !a Modify > accnt. | > | h Hangup call (ha=all) | s Subscribe presence | rr > (Re-)register | > | H Hold call | u Unsubscribe presence | ru > Unregister | > | v re-inVite (release hold) | t ToGgle Online status | > Cycle > next ac.| > | U send UPDATE | T Set online status | < Cycle > prev ac.| > | ],[ Select next/prev call > +--------------------------+-------------------+ > | x Xfer call | Media Commands: | Status & > Config: | > | X Xfer with Replaces | | > | > | # Send RFC 2833 DTMF | cl List ports | d Dump > status | > | * Send DTMF with INFO | cc Connect port | dd Dump > detailed | > | dq Dump curr. call quality | cd Disconnect port | dc Dump > config | > | | V Adjust audio Volume | f Save > config | > | S Send arbitrary REQUEST | Cp Codec priorities | > | > +-----------------------------------------------------------------------------+ > | q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT > type | > +=============================================================================+ > You have 0 active call > >>> cc 0 0 > 23:36:43.839 pjsua_aud.c Conf connect: 0 --> 0 > 23:36:43.843 pjsua_aud.c .Set sound device: capture=-1, playback=-2 > 23:36:43.848 pjsua_app.c ..Turning sound device ON > 23:36:43.851 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms > 23:36:43.964 wmme_dev.c ... WaveAPI Sound player "Wave mapper" > initialized (format=PCM, clock_rate=16000, channel_count=1, > samples_per_frame=320 (20ms)) > 23:36:44.006 wmme_dev.c ... WaveAPI Sound recorder "Wave mapper" > initialized (format=PCM, clock_rate=16000, channel_count=1, > samples_per_frame=320 (20ms)) > 23:36:44.015 ec009E5FC0 ...AEC created, clock_rate=16000, > channel=1, samples per frame=320, tail length=200 ms, latency=0 ms > 23:36:44.022 wmme_dev.c ...WMME playback stream started > 23:36:44.027 wmme_dev.c ...WMME capture stream started > 23:36:44.041 conference.c .Port 0 (Wave mapper) transmitting to > port 0 (Wave mapper) > Success > >>> 23:36:44.070 Master/sound !Underflow, buf_cnt=0, will generate 1 > frame > > 2016-03-12 15:28 GMT+02:00 Bill Gardner <billg at wavearts.com > <mailto:billg at wavearts.com>>: > > The pjsua command cc 0 0 routes the mic audio back to the speaker, > hence there is no codec involved. If this sounds bad then it must > be an issue with the audio device or choice of > portaudio/mme/wasapi drivers. I would try a different audio device > (plug in a USB headset) and look at your audio settings.If you > continue to have problems please post a logfile. > > Regards, > > Bill > > > On 3/12/2016 5:32 AM, Ebubekir DEM?R wrote: >> Hi, >> Thanks for reply. >> However, I changed codec priority with Cp command in pjsua, I can >> not find quality sound. >> Is it a problem in my computer's audio settings? In addition , >> Should I change config_site.h file? >> >> Best regards. >> >> 2016-03-12 0:43 GMT+02:00 David Villasmil Govea >> <david.villasmil at gmail.com <mailto:david.villasmil at gmail.com>>: >> >> What codec are you using? You could try a different codec. >> >> On Fri, Mar 11, 2016 at 11:40 PM Ebubekir DEM?R >> <dmrebubekir at gmail.com <mailto:dmrebubekir at gmail.com>> wrote: >> >> Hi, >> >> I am trying to pjsua in windows 10.It was compiled >> using visual studio express 2012.my config_site.h is >> below. But I hear metallic voice when run cc 0 0 in pjsua >> app.What should I do for improving the sound quality? >> config_site.h: >> #include <pj/config_site_sample.h> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> >> >> _______________________________________________ >> Visit our blog:http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... 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