Hi Ken, Perhaps you should switch the active call first, check ']'/'[' command. BR, nanang On Fri, Jul 27, 2012 at 12:22 AM, Ken Resander <kresander at yahoo.com> wrote: > > I am new to pjsip and need to learn how to use video & audio step-by-step > by using local loopback. Friends and relatives would not know how to be a > remote test site of course and cannot assist. > > I tried: pjsua --video --vcapture-dev=-1 -vrender-dev=-1 > sip:127.0.0.1:5060 > > Q1. Is the command above the right way? > > Q2. If the command is correct, what else might be the problem? (console > log > attached) > > I expected the web-camera (a Logitech) to pick me up me sitting in front > of the computer and transfer video and sound (mic built into to the webcam) > to the same local computer. Then as an incoming call. I could hear the ring > tone, but user-interface command-letter 'a' (answer call) did not have any > effect. Could only enter letter q to quit. Also have tried local loopback > with the --play-avi=AVIFILE option with the same result (started a thread > describing this). The webcamera works fine with Cheese and Skype. > > > Q3. sip:music at iptel.org plays music. Are there other sip URIs that > provide video&audio in a similar fashion? Would be useful for learning and > testing pjsip video. > > > A bit long maybe, but here is the console log text: > > ken at meijin-desktop:~$ > ./pjs201c/pjproject-2.0.1/pjsip-apps/bin/pjsua-i686-pc-linux-gnu --video > --vcapture-dev=-1 --vrender-dev=-1 sip:127.0.0.1:5060 > 12:36:17.591 os_core_unix.c !pjlib 2.0.1 for POSIX initialized > 12:36:17.592 sip_endpoint.c .Creating endpoint instance... > 12:36:17.593 pjlib .select() I/O Queue created (0xa6c9170) > 12:36:17.593 sip_endpoint.c .Module "mod-msg-print" registered > 12:36:17.593 sip_transport. .Transport manager created. > 12:36:17.593 pjsua_core.c .PJSUA state changed: NULL --> CREATED > 12:36:17.593 sip_endpoint.c .Module "mod-pjsua-log" registered > 12:36:17.593 sip_endpoint.c .Module "mod-tsx-layer" registered > 12:36:17.593 sip_endpoint.c .Module "mod-stateful-util" registered > 12:36:17.593 sip_endpoint.c .Module "mod-ua" registered > 12:36:17.593 sip_endpoint.c .Module "mod-100rel" registered > 12:36:17.593 sip_endpoint.c .Module "mod-pjsua" registered > 12:36:17.593 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) > 12:36:17.666 pa_dev.c ..PortAudio sound library initialized, > status=0 > 12:36:17.666 pa_dev.c ..PortAudio host api count=2 > 12:36:17.666 pa_dev.c ..Sound device count=10 > 12:36:17.666 pjlib ..select() I/O Queue created (0xa6e8bd4) > 12:36:17.696 pjsua_vid.c ..Initializing video subsystem.. > 12:36:17.698 ffmpeg_vid_cod ...Cannot find H264 encoder in ffmpeg library > 12:36:17.698 colorbar_dev.c ...Colorbar video src initialized with 1 > device(s): > 12:36:17.698 colorbar_dev.c ... 0: Colorbar generator > 12:36:17.772 sdl_dev.c ...SDL 2.0 initialized > 12:36:17.772 sip_endpoint.c .Module "mod-evsub" registered > 12:36:17.772 sip_endpoint.c .Module "mod-presence" registered > 12:36:17.772 sip_endpoint.c .Module "mod-mwi" registered > 12:36:17.772 sip_endpoint.c .Module "mod-refer" registered > 12:36:17.772 sip_endpoint.c .Module "mod-pjsua-pres" registered > 12:36:17.772 sip_endpoint.c .Module "mod-pjsua-im" registered > 12:36:17.772 sip_endpoint.c .Module "mod-pjsua-options" registered > 12:36:17.773 pjsua_core.c .1 SIP worker threads created > 12:36:17.773 pjsua_core.c .pjsua version 2.0.1 for > Linux-2.6.32.41/i686/glibc-2.11 initialized > 12:36:17.773 pjsua_core.c .PJSUA state changed: CREATED --> INIT > 12:36:17.773 sip_endpoint.c Module "mod-default-handler" registered > 12:36:17.773 pjsua_core.c SIP UDP socket reachable at 192.168.0.2:5060 > 12:36:17.773 udp0xa7111e0 SIP UDP transport started, published address > is 192.168.0.2:5060 > 12:36:17.773 pjsua_acc.c Adding account: id=<sip:192.168.0.2:5060> > 12:36:17.773 pjsua_acc.c .Account <sip:192.168.0.2:5060> added with id > 0 > 12:36:17.773 pjsua_acc.c Modifying accunt 0 > 12:36:17.773 pjsua_acc.c Acc 0: setting online status to 1.. > 12:36:17.774 tcplis:5060 SIP TCP listener ready for incoming > connections at 192.168.0.2:5060 > 12:36:17.774 pjsua_acc.c Adding account: > id=<sip:192.168.0.2:5060;transport=TCP> > 12:36:17.774 pjsua_acc.c .Account <sip:192.168.0.2:5060;transport=TCP> > added with id 1 > 12:36:17.774 pjsua_acc.c Modifying accunt 1 > 12:36:17.774 pjsua_acc.c Acc 1: setting online status to 1.. > 12:36:17.774 pjsua_pres.c Adding buddy: sip:127.0.0.1:5060 > 12:36:17.774 pjsua_pres.c .Buddy 0 added. > 12:36:17.774 pjsua_pres.c .Buddy 0: unsubscribing presence.. > 12:36:17.774 pjsua_pres.c ..Buddy 0: updating presence.. > 12:36:17.774 pjsua_core.c PJSUA state changed: INIT --> STARTING > 12:36:17.774 pjsua_media.c ..NAT type detection failed: Invalid STUN > server or server not configured (PJNATH_ESTUNINSERVER) > 12:36:17.774 sip_endpoint.c .Module "mod-unsolicited-mwi" registered > 12:36:17.774 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING > 12:36:17.774 pjsua_call.c Making call with acc #1 to sip:127.0.0.1:5060 > 12:36:17.774 pjsua_aud.c .Set sound device: capture=-1, playback=-2 > 12:36:17.774 pjsua_app.c ..Turning sound device ON > 12:36:17.774 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms > 12:36:17.777 ec0xa7304e8 ...AEC created, clock_rate=16000, channel=1, > samples per frame=320, tail length=200 ms, latency=100 ms > 12:36:17.777 pjsua_media.c .Call 0: initializing media.. > 12:36:17.777 pjsua_media.c ..RTP socket reachable at 192.168.0.2:40000 > 12:36:17.777 pjsua_media.c ..RTCP socket reachable at 192.168.0.2:40001 > 12:36:17.778 pjsua_media.c ..RTP socket reachable at 192.168.0.2:40002 > 12:36:17.778 pjsua_media.c ..RTCP socket reachable at 192.168.0.2:40003 > 12:36:17.778 pjsua_media.c ..Media index 0 selected for audio call 0 > 12:36:17.779 pjsua_core.c ....TX 1266 bytes Request msg INVITE/cseq=634 > (tdta0xaa6cae0) to UDP 127.0.0.1:5060: > INVITE sip:127.0.0.1:5060 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.0.2:5060;rport;branch=z9hG4bKPjt-FyKRLPJg5xEJpfGH8jOodfbKsHXt0d > Max-Forwards: 70 > From: <sip:192.168.0.2>;tag=knD2N3cRRPB8JbWQchIdnohvz4BLCu-j > To: sip:127.0.0.1 > Contact: <sip:192.168.0.2:5060;ob> > Call-ID: 7q3ZVMr2iBByiJVjO6qULHQWBriOBsUh > CSeq: 634 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Supported: replaces, 100rel, timer, norefersub > Session-Expires: 1800 > Min-SE: 90 > User-Agent: PJSUA v2.0.1 Linux-2.6.32.41/i686/glibc-2.11 > Content-Type: application/sdp > Content-Length: 656 > > v=0 > o=- 3552291377 3552291377 IN IP4 192.168.0.2 > s=pjmedia > c=IN IP4 192.168.0.2 > b=AS:352 > t=0 0 > a=X-nat:0 > m=audio 40000 RTP/AVP 98 97 99 104 3 0 8 9 96 > c=IN IP4 192.168.0.2 > b=TIAS:64000 > a=rtcp:40001 IN IP4 192.168.0.2 > 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 > m=video 40002 RTP/AVP 96 > c=IN IP4 192.168.0.2 > b=TIAS:256000 > a=rtcp:40003 IN IP4 192.168.0.2 > a=sendrecv > a=rtpmap:96 H263-1998/90000 > a=fmtp:96 CIF=1;QCIF=1 > > --end msg-- > 12:36:17.779 pjsua_core.c .RX 1266 bytes Request msg INVITE/cseq=634 > (rdata0xa72c1f4) from UDP 127.0.0.1:5060: > INVITE sip:127.0.0.1:5060 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.0.2:5060;rport;branch=z9hG4bKPjt-FyKRLPJg5xEJpfGH8jOodfbKsHXt0d > Max-Forwards: 70 > From: <sip:192.168.0.2>;tag=knD2N3cRRPB8JbWQchIdnohvz4BLCu-j > To: sip:127.0.0.1 > Contact: <sip:192.168.0.2:5060;ob> > Call-ID: 7q3ZVMr2iBByiJVjO6qULHQWBriOBsUh > CSeq: 634 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Supported: replaces, 100rel, timer, norefersub > Session-Expires: 1800 > Min-SE: 90 > User-Agent: PJSUA v2.0.1 Linux-2.6.32.41/i686/glibc-2.11 > Content-Type: application/sdp > Content-Length: 656 > > v=0 > o=- 3552291377 3552291377 IN IP4 192.168.0.2 > s=pjmedia > c=IN IP4 192.168.0.2 > b=AS:352 > t=0 0 > a=X-nat:0 > m=audio 40000 RTP/AVP 98 97 99 104 3 0 8 9 96 > c=IN IP4 192.168.0.2 > b=TIAS:64000 > a=rtcp:40001 IN IP4 192.168.0.2 > 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 > m=video 40002 RTP/AVP 96 > c=IN IP4 192.168.0.2 > b=TIAS:256000 > a=rtcp:40003 IN IP4 192.168.0.2 > a=sendrecv > a=rtpmap:96 H263-1998/90000 > a=fmtp:96 CIF=1;QCIF=1 > > --end msg-- > 12:36:17.779 pjsua_app.c .......Call 0 state changed to CALLING > 12:36:17.779 pjsua_call.c !.Incoming Request msg INVITE/cseq=634 > (rdata0xa72c1f4) > 12:36:17.779 pjsua_media.c ..Call 1: initializing media.. > 12:36:17.780 pjsua_media.c ...RTP socket reachable at 192.168.0.2:40004 > 12:36:17.780 pjsua_media.c ...RTCP socket reachable at 192.168.0.2:40005 > 12:36:17.780 pjsua_media.c ...RTP socket reachable at 192.168.0.2:40006 > 12:36:17.780 pjsua_media.c ...RTCP socket reachable at 192.168.0.2:40007 > 12:36:17.780 pjsua_media.c ...Media index 0 selected for audio call 1 > 12:36:17.781 pjsua_core.c .....TX 300 bytes Response msg > 100/INVITE/cseq=634 (tdta0xaa80408) to UDP 127.0.0.1:5060: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.0.2:5060;rport=5060;received=127.0.0.1;branch=z9hG4bKPjt-FyKRLPJg5xEJpfGH8jOodfbKsHXt0d > Call-ID: 7q3ZVMr2iBByiJVjO6qULHQWBriOBsUh > From: <sip:192.168.0.2>;tag=knD2N3cRRPB8JbWQchIdnohvz4BLCu-j > To: <sip:127.0.0.1> > CSeq: 634 INVITE > Content-Length: 0 > > > --end msg-- > 12:36:17.781 pjsua_aud.c ..Conf connect: 2 --> 0 > 12:36:17.781 conference.c ...Port 2 (ring) transmitting to port 0 > (Intel 82801DB-ICH4: Intel 82801DB-ICH4 (hw:0,0)) > 12:36:17.781 pjsua_app.c ..Incoming call for account 0! > Media count: 1 audio & 1 video > To reject the video, type "vid disable" first, before answering the call! > From: <sip:192.168.0.2> > To: <sip:127.0.0.1> > Press a to answer or h to reject call > 12:36:17.781 pjsua_core.c .RX 300 bytes Response msg > 100/INVITE/cseq=634 (rdata0xa72c1f4) from UDP 127.0.0.1:5060: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.0.2:5060;rport=5060;received=127.0.0.1;branch=z9hG4bKPjt-FyKRLPJg5xEJpfGH8jOodfbKsHXt0d > Call-ID: 7q3ZVMr2iBByiJVjO6qULHQWBriOBsUh > From: <sip:192.168.0.2>;tag=knD2N3cRRPB8JbWQchIdnohvz4BLCu-j > To: <sip:127.0.0.1> > CSeq: 634 INVITE > Content-Length: 0 > > > --end msg-- > >>>> > Account list: > [ 0] <sip:192.168.0.2:5060>: does not register > Online status: Online > *[ 1] <sip:192.168.0.2:5060;transport=TCP>: does not register > Online status: Online > Buddy list: > [ 1] <?> sip:127.0.0.1:5060 > > > +=============================================================================+ > | 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 | > | > > +-----------------------------------------------------------------------------+ > | Video: "vid help" for more info > | > > +-----------------------------------------------------------------------------+ > | q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type > | > > +=============================================================================+ > | Video will be enabled in the next offer/answer > | > > +=============================================================================+ > >>> a > No pending incoming call > >>> q > 12:36:26.218 pjsua_core.c !Shutting down, flags=0... > > > _______________________________________________ > 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 >
