Hey Paolo, So I discovered that my issues were from two separate problems. One is that my microphones are dual channel and for some reason aren't working correctly with pjsystest and pjsua, although they work fine with the other demo programs. Still trying to figure that out. The reason that sound playback wasn't working was because pjsua was maxing out the CPU on the ARM board, so nothing was coming through. I wound up setting a bunch of options in the config_site.h to turn off echo cancellation, turn off floating point, and use the audio switchboard instead of conference bridge to get the CPU load down. Then I can hear incoming calls. Try running top when you're running pjsua and see if you're maxing out the CPU, and if so maybe those options will fix your problems. On Thu, Feb 14, 2013 at 3:20 AM, Paolo Pellegatti < paolo.pellegatti at axesstmc.com> wrote: > Hi, > > I have notice the same issue regarding the file .wav missing downloaded > from > web site on .bz2 archive. > > Have you tried to download pjsip 2.x from svn repository ? > > You can use the following command: > > svn co http://svn.pjsip.org/repos/pjproject/trunk pjproject-svn > > At the end of checkout, check your local copy > (./pjproject-svn/tests/pjsua/wavs) if it contains the wav files. > > I have built the pjsip w/ ALSA on ARM machine directly (no cross-compile) > using the following steps and for me it works: > > 1) Check & Install libasound2-dev package > > dpkg -l libasound2-dev > > If not installed use --> apt-get install libasound2-dev > > 2) Configure pjsip 2.x using for example this command: > > CFLAGS="-g -Wno-unused-label" ./configure --disable-floating-point > --disable-speex-aec \ > --disable-l16-codec --disable-gsm-codec --disable-ilbc-codec > --disable-g722-codec \ > --disable-g7221-codec --disable-speex-codec --disable-opencore-amr > --disable-ffmpeg \ > --disable-sdl --enable-resample-dll --disable-large-filter > --disable-video > > 3) Building the project > > make dep && make > > 4) Installing pjsip on local folder at the same level of pjproject-svn > folder project > > mkdir ../build-pjsip/ > make install DESTDIR=$(pwd)/../build-pjsip > mkdir ../build-pjsip/usr/local/bin > cp pjsip-apps/bin/pjs* ../build-pjsip/usr/local/bin/ > ## rename some files > mv ../build-pjsip/usr/local/bin/pjsystest-* > ../build-pjsip/usr/local/bin/pjsystest > mv ../build-pjsip/usr/local/bin/pjsua-* > ../build-pjsip/usr/local/bin/pjsua > ## copy all wav files on the same location of pjsua > cp tests/pjsua/wavs/* ../build-pjsip/usr/local/bin/ > > ## Next create a tar file with all files > cd ../build-pjsip/usr/local > tar -cvzpf ../pjsip-archive.tar.gz * > > Copy pjsip-archive.tar.gz on your target ARM machine root "/" folder and > extract it > > 5) Check the dependencies list with ldd command on pjsua and pjsystest if > libasound.so.2 is present and found > > 6) Execute pjsystest from /usr/bin folder > > My problem is pjsua, it seems doesn't works. > I can register correctly to a SIP server, I can hear a tone of a calling > and > accept it but I didn't hear any sound or noise after that. > Where am I wrong ? > > This is my log of an incoming call: > > 03:18:52.090 pjsua_app.c ..Incoming call for account 2! > Media count: 1 audio & 0 video > From: <sip:101@192.168.1.105> > To: <sip:100 at 192.168.20.71> > Press a to answer or h to reject call > 03:18:52.093 os_core_unix.c Info: possibly re-registering existing thread > a > Answer with code (100-699) (empty to cancel): 200 > 03:19:00.123 pjsua_call.c !Answering call 0: code=200 > 03:19:00.124 pjsua_media.c ...Call 0: updating media.. > 03:19:00.125 pjsua_aud.c ....Audio channel update.. > 03:19:00.125 strm0x23dca4 .....VAD temporarily disabled > 03:19:00.126 strm0x23dca4 .....Encoder stream started > 03:19:00.126 strm0x23dca4 .....Decoder stream started > 03:19:00.128 pjsua_media.c ....Audio updated, stream #0: PCMU (sendrecv) > 03:19:00.130 pjsua_app.c ...Call 0 media 0 [type=audio], status is > Active > 03:19:00.130 pjsua_aud.c ...Conf disconnect: 2 -x- 0 > 03:19:00.131 conference.c ....Port 2 (ring) stop transmitting to port 0 > (am35 > 17evm: (hw:0,0)) > 03:19:00.131 pjsua_aud.c ...Conf connect: 3 --> 0 > 03:19:00.131 conference.c ....Port 3 (sip:101 at 192.168.1.105:5060) > transmittin > g to port 0 (am3517evm: (hw:0,0)) > 03:19:00.131 pjsua_aud.c ...Conf connect: 0 --> 3 > 03:19:00.131 conference.c ....Port 0 (am3517evm: (hw:0,0)) transmitting > to p > ort 3 (sip:101 at 192.168.1.105:5060) > 03:19:00.135 pjsua_core.c !....TX 848 bytes Response msg > 200/INVITE/cseq=1 > (td > ta0x232df0) to TCP 192.168.1.105:5060: > SIP/2.0 200 OK > Via: SIP/2.0/TCP > 192.168.1.105:5060;rport=5060;received=192.168.1.105;branch=z9h > G4bK-d8754z-6a3a592d8c255b2c-1---d8754z- > Call-ID: MjFlM2MyNjg0ZjZhNDBhMWQ2ODRjMTg1ZmJlYjBhNjQ. > From: <sip:101@192.168.1.105>;tag=6906ce60 > To: <sip:100 at 192.168.20.71>;tag=KelLS.BBeKxsYLD7R9XDNalIAp3fyqm6 > CSeq: 1 INVITE > Contact: <sip:100 at 192.168.20.71:5060;transport=TCP;ob> > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAG > E, OPTIONS > Supported: replaces, 100rel, timer, norefersub > Content-Type: application/sdp > Content-Length: 277 > > v=0 > o=- 3161816332 3161816333 IN IP4 192.168.20.71 > s=pjmedia > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 0 101 > c=IN IP4 192.168.20.71 > b=TIAS:64000 > a=rtcp:4001 IN IP4 192.168.20.71 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > --end msg-- > 03:19:00.138 pjsua_app.c .......Call 0 state changed to CONNECTING > >>> 03:19:00.244 tcplis:5060 !TCP listener 192.168.20.71:5060: got > incoming T > CP connection from 192.168.1.105:57075, sock=9 > 03:19:00.244 tcps0x241dec TCP server transport created > 03:19:00.245 pjsua_app.c SIP TCP transport is connected to > [192.168.1.105:57 > 075] > 03:19:00.246 pjsua_core.c .RX 462 bytes Request msg ACK/cseq=1 > (rdata0x241f94 > ) from TCP 192.168.1.105:57075: > ACK sip:100 at 192.168.20.71:5060;transport=TCP;ob SIP/2.0 > Via: SIP/2.0/TCP > 192.168.1.105:5060;branch=z9hG4bK-d8754z-72589c3d6a550955-1---d > 8754z-;rport > Max-Forwards: 70 > Contact: <sip:101 at 192.168.1.105:5060;transport=TCP> > To: <sip:100 at 192.168.20.71>;tag=KelLS.BBeKxsYLD7R9XDNalIAp3fyqm6 > From: <sip:101@192.168.1.105>;tag=6906ce60 > Call-ID: MjFlM2MyNjg0ZjZhNDBhMWQ2ODRjMTg1ZmJlYjBhNjQ. > CSeq: 1 ACK > User-Agent: 3CXPhoneSystem 11.0.25940.0 > Content-Length: 0 > --end msg-- > 03:19:00.249 pjsua_app.c ...Call 0 state changed to CONFIRMED > 03:19:00.761 strm0x23dca4 VAD re-enabled > 03:19:01.247 strm0x23dca4 Bad RTP pt 126 (expecting 0) > 03:19:01.247 strm0x23dca4 Bad RTP pt 126 (expecting 0) > 03:19:01.248 strm0x23dca4 Bad RTP pt 126 (expecting 0) > 03:19:05.140 sound_port.c EC suspended because of inactivity > 03:19:11.558 strm0x23dca4 Bad RTP pt 126 (expecting 0) > 03:19:21.867 strm0x23dca4 Bad RTP pt 126 (expecting 0) > 03:19:32.178 strm0x23dca4 Bad RTP pt 126 (expecting 0) > 03:19:42.489 strm0x23dca4 Bad RTP pt 126 (expecting 0) > 03:19:47.908 pjsua_core.c .RX 462 bytes Request msg BYE/cseq=2 > (rdata0x229a54 > ) from TCP 192.168.1.105:5060: > BYE sip:100 at 192.168.20.71:5060;transport=TCP;ob SIP/2.0 > Via: SIP/2.0/TCP > 192.168.1.105:5060;branch=z9hG4bK-d8754z-5902447b6474ee5a-1---d > 8754z-;rport > Max-Forwards: 70 > Contact: <sip:101 at 192.168.1.105:5060;transport=TCP> > To: <sip:100 at 192.168.20.71>;tag=KelLS.BBeKxsYLD7R9XDNalIAp3fyqm6 > From: <sip:101@192.168.1.105>;tag=6906ce60 > Call-ID: MjFlM2MyNjg0ZjZhNDBhMWQ2ODRjMTg1ZmJlYjBhNjQ. > CSeq: 2 BYE > User-Agent: 3CXPhoneSystem 11.0.25940.0 > Content-Length: 0 > > > --end msg-- > 03:19:47.909 pjsua_core.c .......TX 338 bytes Response msg > 200/BYE/cseq=2 > (td > ta0x232df0) to TCP 192.168.1.105:5060: > SIP/2.0 200 OK > Via: SIP/2.0/TCP > 192.168.1.105:5060;rport=5060;received=192.168.1.105;branch=z9h > G4bK-d8754z-5902447b6474ee5a-1---d8754z- > Call-ID: MjFlM2MyNjg0ZjZhNDBhMWQ2ODRjMTg1ZmJlYjBhNjQ. > From: <sip:101@192.168.1.105>;tag=6906ce60 > To: <sip:100 at 192.168.20.71>;tag=KelLS.BBeKxsYLD7R9XDNalIAp3fyqm6 > CSeq: 2 BYE > Content-Length: 0 > > > --end msg-- > 03:19:47.910 pjsua_app.c ......Call 0 is DISCONNECTED [reason=200 > (Normal ca > ll clearing)] > 03:19:47.915 pjsua_app.c !...... > [DISCONNCTD] To: <sip:101 at 192.168.1.105>;tag=6906ce60 > Call time: 00h:00m:47s, 1st res in 8059 ms, conn in 8182ms > #0 audio PCMU @8kHz, sendrecv, peer=192.168.1.105:5064 > SRTP status: Not active Crypto-suite: (null) > RX pt=0, last update:00h:00m:01.331s ago > total 7pkt 28B (308B +IP hdr) @avg=4bps/51bps > pkt loss=0 (0.0%), discrd=2 (28.6%), dup=2 (28.6%), reord=0 > (0.0%) > (msec) min avg max last dev > loss period: 0.000 0.000 0.000 0.000 0.000 > jitter : -0.001 0.000 0.000 0.000 0.000 > TX pt=0, ptime=20, last update:00h:00m:02.964s ago > total 1.1Kpkt 181.1KB (226.4KB +IP hdr) @avg=30.3Kbps/37.9Kbps > pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) > (msec) min avg max last dev > loss period: 0.000 0.000 0.000 0.000 0.000 > jitter : 1.500 1.924 2.375 1.875 0.227 > RTT msec : 1.815 2.294 3.326 2.105 0.498 > 03:19:47.916 pjsua_media.c ......Call 0: deinitializing media.. > 03:19:47.925 pjsua_media.c ........Media stream call00:0 is destroyed > 03:19:48.925 pjsua_aud.c Closing sound device after idle for 1 > second(s) > 03:19:48.925 pjsua_app.c .Turning sound device OFF > 03:19:48.926 pjsua_aud.c .Closing am3517evm: (hw:0,0) sound playback > device > and am3517evm: (hw:0,0) sound capture device > > > > > > > > > _______________________________________________ > 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 > -- *haomiao 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