Hello *Gaurav, Varun, Dennis, Nishant* Can you please tell me why after installing pjsip 2.1 perfectly, with libasound2 and all , i still do not have audio playback? (i checked with speaker-test, alsa sink, src mplayer, vlc, ffmpeg my speaker and mic is available without pjsip it works, but with pjsip i still do not hear any single audio playback, also when i am connected i have no microphone capture) Please can you kindly share, i have been trying this for about now 4 weeks, still its not working at all. Please see the details of following steps how i installed it and how i tested it. *Step 1*: install and run $ cd /var/tmp $ wget http://www.pjsip.org/release/2.1/pjproject-2.1.tar.bz2 $ tar xvfj pjproject-2.1.tar.bz2 $ cd pjproject-2.1 $ ./configure $ make dep && make && make install # Python enable (optional) $ cd /var/tmp/pjproject-2.1.0/pjsip-apps/src/python $ python setup.py install $ python Python 2.7.5+ (default, Sep 19 2013, 13:48:49) [GCC 4.8.1] on linux2 Type "help", "copyright", "credits" or "license" for more information. >>> import pjsua >>> *Step 2*: Basic kick start sample to register and make call, by manually assigning playback id and capture id , this also do not work for audio capture and playback: https://gist.github.com/anonymous/7768285 Here you can see i used the latest release built in pjsua which also giving no sound and no luck to capture microphone. $ ./pjsua-x86_64-unknown-linux-gnu 13:24:33.632 os_core_unix.c !pjlib 2.1 for POSIX initialized 13:24:33.632 sip_endpoint.c .Creating endpoint instance... 13:24:33.633 pjlib .select() I/O Queue created (0x20fb8a0) 13:24:33.633 sip_endpoint.c .Module "mod-msg-print" registered 13:24:33.633 sip_transport. .Transport manager created. 13:24:33.633 pjsua_core.c .PJSUA state changed: NULL --> CREATED 13:24:33.633 sip_endpoint.c .Module "mod-pjsua-log" registered 13:24:33.633 sip_endpoint.c .Module "mod-tsx-layer" registered 13:24:33.633 sip_endpoint.c .Module "mod-stateful-util" registered 13:24:33.633 sip_endpoint.c .Module "mod-ua" registered 13:24:33.633 sip_endpoint.c .Module "mod-100rel" registered 13:24:33.633 sip_endpoint.c .Module "mod-pjsua" registered 13:24:33.633 sip_endpoint.c .Module "mod-invite" registered bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) 13:24:33.702 pa_dev.c ..PortAudio sound library initialized, status=0 13:24:33.702 pa_dev.c ..PortAudio host api count=2 13:24:33.702 pa_dev.c ..Sound device count=20 13:24:33.702 pjlib ..select() I/O Queue created (0x21579f8) 13:24:33.711 sip_endpoint.c .Module "mod-evsub" registered 13:24:33.711 sip_endpoint.c .Module "mod-presence" registered 13:24:33.711 sip_endpoint.c .Module "mod-mwi" registered 13:24:33.711 sip_endpoint.c .Module "mod-refer" registered 13:24:33.711 sip_endpoint.c .Module "mod-pjsua-pres" registered 13:24:33.711 sip_endpoint.c .Module "mod-pjsua-im" registered 13:24:33.711 sip_endpoint.c .Module "mod-pjsua-options" registered 13:24:33.711 pjsua_core.c .1 SIP worker threads created 13:24:33.711 pjsua_core.c .pjsua version 2.1 for Linux-3.11.0.12/x86_64/ glibc-2.17 initialized 13:24:33.711 pjsua_core.c .PJSUA state changed: CREATED --> INIT 13:24:33.711 sip_endpoint.c Module "mod-default-handler" registered 13:24:33.711 pjsua_core.c bind() error: Address already in use [status= 120098] 13:24:33.711 pjsua_core.c Shutting down, flags=0... 13:24:33.711 pjsua_core.c PJSUA state changed: INIT --> CLOSING 13:24:33.721 pjsua_call.c .Hangup all calls.. 13:24:33.721 pjsua_pres.c .Shutting down presence.. 13:24:33.721 pjsua_media.c .Shutting down media.. 13:24:33.721 pjsua_media.c ..Call 0: deinitializing media.. 13:24:33.721 pjsua_media.c ..Call 1: deinitializing media.. 13:24:33.721 pjsua_media.c ..Call 2: deinitializing media.. 13:24:33.721 pjsua_media.c ..Call 3: deinitializing media.. 13:24:34.203 pa_dev.c ..PortAudio sound library shutting down.. 13:24:35.210 pjsua_core.c .Destroying... 13:24:35.210 sip_transactio .Stopping transaction layer module 13:24:35.210 sip_transactio .Stopped transaction layer module 13:24:35.210 sip_endpoint.c .Module "mod-default-handler" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-pjsua-options" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-pjsua-im" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-pjsua-pres" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-pjsua" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-stateful-util" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-refer" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-mwi" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-presence" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-evsub" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-invite" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-100rel" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-ua" unregistered 13:24:35.210 sip_transactio .Transaction layer module destroyed 13:24:35.210 sip_endpoint.c .Module "mod-tsx-layer" unregistered 13:24:35.210 sip_endpoint.c .Module "mod-msg-print" unregistered 13:24:35.211 sip_endpoint.c .Module "mod-pjsua-log" unregistered 13:24:35.211 sip_endpoint.c .Endpoint 0x20f0b08 destroyed 13:24:35.211 pjsua_core.c .PJSUA state changed: CLOSING --> NULL 13:24:35.211 pjsua_core.c .PJSUA destroyed... sun at sun-Alienware-X51:/var/tmp/pjproject-2.1.0/pjsip-apps/bin$ ./pjsua- x86_64-unknown-linux-gnu 13:24:51.994 os_core_unix.c !pjlib 2.1 for POSIX initialized 13:24:51.995 sip_endpoint.c .Creating endpoint instance... 13:24:51.995 pjlib .select() I/O Queue created (0x9d98a0) 13:24:51.995 sip_endpoint.c .Module "mod-msg-print" registered 13:24:51.995 sip_transport. .Transport manager created. 13:24:51.995 pjsua_core.c .PJSUA state changed: NULL --> CREATED 13:24:51.995 sip_endpoint.c .Module "mod-pjsua-log" registered 13:24:51.995 sip_endpoint.c .Module "mod-tsx-layer" registered 13:24:51.995 sip_endpoint.c .Module "mod-stateful-util" registered 13:24:51.995 sip_endpoint.c .Module "mod-ua" registered 13:24:51.995 sip_endpoint.c .Module "mod-100rel" registered 13:24:51.995 sip_endpoint.c .Module "mod-pjsua" registered 13:24:51.995 sip_endpoint.c .Module "mod-invite" registered bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) 13:24:52.013 pa_dev.c ..PortAudio sound library initialized, status=0 13:24:52.013 pa_dev.c ..PortAudio host api count=2 13:24:52.013 pa_dev.c ..Sound device count=20 13:24:52.013 pjlib ..select() I/O Queue created (0xa359f8) 13:24:52.016 sip_endpoint.c .Module "mod-evsub" registered 13:24:52.016 sip_endpoint.c .Module "mod-presence" registered 13:24:52.017 sip_endpoint.c .Module "mod-mwi" registered 13:24:52.017 sip_endpoint.c .Module "mod-refer" registered 13:24:52.017 sip_endpoint.c .Module "mod-pjsua-pres" registered 13:24:52.017 sip_endpoint.c .Module "mod-pjsua-im" registered 13:24:52.017 sip_endpoint.c .Module "mod-pjsua-options" registered 13:24:52.017 pjsua_core.c .1 SIP worker threads created 13:24:52.017 pjsua_core.c .pjsua version 2.1 for Linux-3.11.0.12/x86_64/ glibc-2.17 initialized 13:24:52.017 pjsua_core.c .PJSUA state changed: CREATED --> INIT 13:24:52.017 sip_endpoint.c Module "mod-default-handler" registered 13:24:52.017 pjsua_core.c SIP UDP socket reachable at 192.168.1.19:5060 13:24:52.017 udp0xa4e6e0 SIP UDP transport started, published address is 192.168.1.19:5060 13:24:52.017 pjsua_acc.c Adding account: id=<sip:192.168.1.19:5060> 13:24:52.017 pjsua_acc.c .Account <sip:192.168.1.19:5060> added with id 0 13:24:52.017 pjsua_acc.c Acc 0: setting online status to 1.. 13:24:52.017 tcplis:5060 SIP TCP listener ready for incoming connections at 192.168.1.19:5060 13:24:52.017 pjsua_acc.c Adding account: id=<sip:192.168.1.19:5060; transport=TCP> 13:24:52.017 pjsua_acc.c .Account <sip:192.168.1.19:5060;transport=TCP> added with id 1 13:24:52.017 pjsua_acc.c Acc 1: setting online status to 1.. 13:24:52.017 pjsua_core.c PJSUA state changed: INIT --> STARTING 13:24:52.017 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 13:24:52.017 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING >>>> Account list: [ 0] <sip:192.168.1.19:5060>: does not register Online status: Online *[ 1] <sip:192.168.1.19:5060;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 >>> m (You currently have 0 calls) Buddy list: -none- Choices: 0 For current dialog. -1 All 0 buddies in buddy list [1 - 0] Select from buddy list URL An URL <Enter> Empty input (or 'q') to cancel Make call: sip:9198 at 192.168.1.12 13:25:48.064 pjsua_call.c Making call with acc #1 to sip:9198 at 192.168.1.12 13:25:48.065 pjsua_aud.c .Set sound device: capture=-1, playback=-2 13:25:48.065 pjsua_app.c ..Turning sound device ON 13:25:48.065 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994 Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994 13:25:48.065 pjsua_app.c ..Turning sound device ON 13:25:48.065 pjsua_aud.c ..Opening sound device PCM at 44100/1/20ms 13:25:48.122 ec0x9fec00 ...AEC created, clock_rate=44100, channel=1, samples per frame=882, tail length=200 ms, latency=100 ms 13:25:48.123 pjsua_media.c .Call 0: initializing media.. 13:25:48.123 pjsua_media.c ..RTP socket reachable at 192.168.1.19:40000 13:25:48.123 pjsua_media.c ..RTCP socket reachable at 192.168.1.19:40001 13:25:48.123 pjsua_media.c ..Media index 0 selected for audio call 0 13:25:48.123 pjsua_core.c ....TX 1107 bytes Request msg INVITE/cseq=18614 (tdta0xadcbd0) to UDP 192.168.1.12:5060: INVITE sip:9198 at 192.168.1.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch= z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig Max-Forwards: 70 From: <sip:192.168.1.19>;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma To: sip:9198 at 192.168.1.12 Contact: <sip:192.168.1.19:5060;ob> Call-ID: aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D CSeq: 18614 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17 Content-Type: application/sdp Content-Length: 475 v=0 o=- 3595062348 3595062348 IN IP4 192.168.1.19 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 40000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.1.19 b=TIAS:64000 a=rtcp:40001 IN IP4 192.168.1.19 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 --end msg-- 13:25:48.123 pjsua_app.c .......Call 0 state changed to CALLING >>> 13:25:48.124 pjsua_core.c .RX 365 bytes Response msg 100/INVITE/cseq= 18614 (rdata0xa4fd48) from UDP 192.168.1.12:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.19:5060;rport=5060;branch= z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig From: <sip:192.168.1.19>;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma To: <sip:9198 at 192.168.1.12> Call-ID: aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D CSeq: 18614 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f Content-Length: 0 --end msg-- 13:25:48.144 os_core_unix.c Info: possibly re-registering existing thread 13:25:48.145 pjsua_core.c .RX 882 bytes Response msg 407/INVITE/cseq= 18614 (rdata0x7f5b40002998) from UDP 192.168.1.12:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.19:5060;rport=5060;branch= z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig From: <sip:192.168.1.19>;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma To: <sip:9198 at 192.168.1.12>;tag=DZ4am8m4t08Xr Call-ID: aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D CSeq: 18614 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="192.168.1.19", nonce= "db2a6c3c-5c29-11e3-a388-3586b66a1730", algorithm=MD5, qop="auth" Content-Length: 0 --end msg-- 13:25:48.145 pjsua_core.c ..TX 334 bytes Request msg ACK/cseq=18614 ( tdta0x7f5b400008c0) to UDP 192.168.1.12:5060: ACK sip:9198 at 192.168.1.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch= z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig Max-Forwards: 70 From: <sip:192.168.1.19>;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma To: sip:9198 at 192.168.1.12;tag=DZ4am8m4t08Xr Call-ID: aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D CSeq: 18614 ACK Content-Length: 0 --end msg-- 13:25:48.145 sip_auth_clien ....Unable to set auth for tdta0xadcbd0: can not find credential for 192.168.1.19/Digest 13:25:48.145 pjsua_app.c .....Call 0 is DISCONNECTED [reason=407 (Proxy Authentication Required)] 13:25:48.145 pjsua_app.c ..... [DISCONNCTD] To: sip:9198 at 192.168.1.12 Call time: 00h:00m:00s, 1st res in 23 ms, conn in 0ms 13:25:48.145 pjsua_media.c .....Call 0: deinitializing media.. 13:25:49.146 pjsua_aud.c !Closing sound device after idle for 1 second(s) 13:25:49.146 pjsua_app.c .Turning sound device OFF 13:25:49.146 pjsua_aud.c .Closing HDA Intel PCH: ALC892 Analog (hw:0,0) sound playback device and HDA Intel PCH: ALC892 Analog (hw:0,0) sound capture device q 13:26:32.391 pjsua_core.c !Shutting down, flags=0... 13:26:32.391 pjsua_core.c PJSUA state changed: RUNNING --> CLOSING 13:26:32.396 pjsua_call.c .Hangup all calls.. 13:26:32.396 pjsua_pres.c .Shutting down presence.. 13:26:32.396 pjsua_media.c .Shutting down media.. 13:26:32.396 pjsua_media.c ..Call 0: deinitializing media.. 13:26:32.396 pjsua_media.c ..Call 1: deinitializing media.. 13:26:32.396 pjsua_media.c ..Call 2: deinitializing media.. 13:26:32.396 pjsua_media.c ..Call 3: deinitializing media.. 13:26:32.524 pa_dev.c ..PortAudio sound library shutting down.. 13:26:33.532 pjsua_core.c .Destroying... 13:26:33.532 sip_transactio .Stopping transaction layer module 13:26:33.532 sip_transactio .Stopped transaction layer module 13:26:33.532 sip_endpoint.c .Module "mod-default-handler" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-unsolicited-mwi" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-pjsua-options" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-pjsua-im" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-pjsua-pres" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-pjsua" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-stateful-util" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-refer" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-mwi" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-presence" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-evsub" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-invite" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-100rel" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-ua" unregistered 13:26:33.532 sip_transactio .Transaction layer module destroyed 13:26:33.532 sip_endpoint.c .Module "mod-tsx-layer" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-msg-print" unregistered 13:26:33.532 sip_endpoint.c .Module "mod-pjsua-log" unregistered 13:26:33.533 tcplis:5060 .SIP TCP listener destroyed 13:26:33.533 sip_endpoint.c .Endpoint 0x9ceb08 destroyed 13:26:33.533 pjsua_core.c .PJSUA state changed: CLOSING --> NULL 13:26:33.533 pjsua_core.c .PJSUA destroyed... Thank you Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20131203/0e85993e/attachment-0001.html>
