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
