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Hello I am looking to setup this scenario you discribed below and i am struggling to make it work Did you find a solution for it ? If it is the case, is it possible to share a sample of working code you found ? Thanks in advance. Regards =========================================================== I am trying to learn how to use pjsip with video with pjsip 2.0.1. I don't have a peer/partner to send to so need to use loopback on the PC that runs Ubuntu 10.04 LTS. I am using pjsua like this: pjsua --play-avi=Path_avi_file sip:127.0.0.1 but it is not doing what I imagined it would do. The avi file atworkpcmmulaw8000.avi contains: video: 320x240 ffmpeg mpeg-4 30fps audio: PCM mulaw mono sampling rate 8000 I converted this from a wmv file found at http://www.funny-games.biz/videos/ in section Office Videos under Bored at Work by the ffmpeg utility. There are details about this in another thread 'Cannot get AVI file to play on pjsip 2.0.1' that I will go back to later. Running the command-line command produced a ringtone, but there were points before the ring that I could not understand. 1. I have read PCM mulaw at 8000Hz is widely supported, but the following lines: "18:09:52.335 avi_player.c Unsupported audio stream 18:09:52.335 avi_player.c AVI file player '/home/ken/pjs201c/pjproject-2.0.1/pjsip-apps/bin/atworkpcmmulaw8000.avi' created with 1 media ports 18:09:52.339 pjsua_app.c AVI player atworkpcmmulaw8000.avi created, dev_id=2 18:09:52.339 pjsua_app.c AVI atworkpcmmulaw8000.avi: audio ignored, format=MPG4" seem to indicate it is not supported. Also I am puzzled by 'audio ignored, format=MPG4' since I did not request mpg4 when converting the audio from wmv to avi. 2. The sip INVITE contains SDP that bears no relationship to the audio in the AVI file given in --play-avi option. I imagined pjsip would inspect the avi header and build a corresponding SDP offer because that is what the offerer has got to offer. Can I tell the SIP INVITE to build SDP using the codec info in the avi file from the command-line? Or more generally, how to specify what goes into the SDP part from the command line? 3. Why is the port 0 for video media (video 0 RTP/AVP 31)? since video is needed... 4. Immediately after the UI menu is a message "You have 2 active calls". Why is this? since I have only made one call... 5. a bit further down I can see the message "No pending calls", but I expected one since it was ringing. I tried command-letter a to answer the call, but this did not have any effect. So quit (q) at this point. 6. I also tried some other wmv-to-avi conversions. For example, h.263 for video and audio pcm_s16le, mono,8000hz. Then both video and audio were reported as not supported. Also mystified by this. I really need to understand this a bit better. Hoping for advice. Ken P.S. Here is the console log (probably longer than it needs to be, but could not decide what to leave out): ken at meijin-desktop:~$ ./pjs201c/pjproject-2.0.1/pjsip-apps/bin/pjsua-i686-pc-linux-gnu --play-avi=/home/ken/pjs201c/pjproject-2.0.1/pjsip-apps/bin/atworkpcmmulaw8000.avi sip:127.0.0.1 18:09:52.129 os_core_unix.c !pjlib 2.0.1 for POSIX initialized 18:09:52.131 sip_endpoint.c .Creating endpoint instance... 18:09:52.131 pjlib .select() I/O Queue created (0xa046170) 18:09:52.131 sip_endpoint.c .Module "mod-msg-print" registered 18:09:52.131 sip_transport. .Transport manager created. 18:09:52.131 pjsua_core.c .PJSUA state changed: NULL --> CREATED 18:09:52.131 sip_endpoint.c .Module "mod-pjsua-log" registered 18:09:52.131 sip_endpoint.c .Module "mod-tsx-layer" registered 18:09:52.131 sip_endpoint.c .Module "mod-stateful-util" registered 18:09:52.131 sip_endpoint.c .Module "mod-ua" registered 18:09:52.131 sip_endpoint.c .Module "mod-100rel" registered 18:09:52.131 sip_endpoint.c .Module "mod-pjsua" registered 18:09:52.132 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) 18:09:52.207 pa_dev.c ..PortAudio sound library initialized, status=0 18:09:52.207 pa_dev.c ..PortAudio host api count=2 18:09:52.207 pa_dev.c ..Sound device count=10 18:09:52.207 pjlib ..select() I/O Queue created (0xa065bd4) 18:09:52.256 pjsua_vid.c ..Initializing video subsystem.. 18:09:52.258 ffmpeg_vid_cod ...Cannot find H264 encoder in ffmpeg library 18:09:52.258 colorbar_dev.c ...Colorbar video src initialized with 1 device(s): 18:09:52.258 colorbar_dev.c ... 0: Colorbar generator 18:09:52.335 sdl_dev.c ...SDL 2.0 initialized 18:09:52.335 sip_endpoint.c .Module "mod-evsub" registered 18:09:52.335 sip_endpoint.c .Module "mod-presence" registered 18:09:52.335 sip_endpoint.c .Module "mod-mwi" registered 18:09:52.335 sip_endpoint.c .Module "mod-refer" registered 18:09:52.335 sip_endpoint.c .Module "mod-pjsua-pres" registered 18:09:52.335 sip_endpoint.c .Module "mod-pjsua-im" registered 18:09:52.335 sip_endpoint.c .Module "mod-pjsua-options" registered 18:09:52.335 pjsua_core.c .1 SIP worker threads created 18:09:52.335 pjsua_core.c .pjsua version 2.0.1 for Linux-2.6.32.41/i686/glibc-2.11 initialized 18:09:52.335 pjsua_core.c .PJSUA state changed: CREATED --> INIT 18:09:52.335 sip_endpoint.c Module "mod-default-handler" registered 18:09:52.335 avi_dev.c AVI dev factory created with 1 virtual device(s) 18:09:52.335 avi_player.c Unsupported audio stream 18:09:52.335 avi_player.c AVI file player '/home/ken/pjs201c/pjproject-2.0.1/pjsip-apps/bin/atworkpcmmulaw8000.avi' created with 1 media ports 18:09:52.339 pjsua_app.c AVI player atworkpcmmulaw8000.avi created, dev_id=2 18:09:52.339 pjsua_app.c AVI atworkpcmmulaw8000.avi: audio ignored, format=MPG4 18:09:52.340 pjsua_core.c SIP UDP socket reachable at 192.168.0.2:5060 18:09:52.340 udp0xa097190 SIP UDP transport started, published address is 192.168.0.2:5060 18:09:52.340 pjsua_acc.c Adding account: id=<sip:192.168.0.2:5060> 18:09:52.340 pjsua_acc.c .Account <sip:192.168.0.2:5060> added with id 0 18:09:52.340 pjsua_acc.c Modifying accunt 0 18:09:52.340 pjsua_acc.c Acc 0: setting online status to 1.. 18:09:52.341 tcplis:5060 SIP TCP listener ready for incoming connections at 192.168.0.2:5060 18:09:52.341 pjsua_acc.c Adding account: id=<sip:192.168.0.2:5060;transport=TCP> 18:09:52.341 pjsua_acc.c .Account <sip:192.168.0.2:5060;transport=TCP> added with id 1 18:09:52.341 pjsua_acc.c Modifying accunt 1 18:09:52.341 pjsua_acc.c Acc 1: setting online status to 1.. 18:09:52.341 pjsua_pres.c Adding buddy: sip:127.0.0.1 18:09:52.341 pjsua_pres.c .Buddy 0 added. 18:09:52.341 pjsua_pres.c .Buddy 0: unsubscribing presence.. 18:09:52.341 pjsua_pres.c ..Buddy 0: updating presence.. 18:09:52.341 pjsua_core.c PJSUA state changed: INIT --> STARTING 18:09:52.341 pjsua_media.c ..NAT type detection failed: Invalid STUN server or server not configured (PJNATH_ESTUNINSERVER) 18:09:52.341 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 18:09:52.341 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING 18:09:52.341 pjsua_call.c Making call with acc #1 to sip:127.0.0.1 18:09:52.341 pjsua_aud.c .Set sound device: capture=-1, playback=-2 18:09:52.341 pjsua_app.c ..Turning sound device ON 18:09:52.341 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms 18:09:52.343 ec0xa0cd710 ...AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=100 ms 18:09:52.344 pjsua_media.c .Call 0: initializing media.. 18:09:52.344 pjsua_media.c ..RTP socket reachable at 192.168.0.2:40000 18:09:52.344 pjsua_media.c ..RTCP socket reachable at 192.168.0.2:40001 18:09:52.344 pjsua_media.c ..Media index 0 selected for audio call 0 18:09:52.345 pjsua_core.c ....TX 1143 bytes Request msg INVITE/cseq=13130 (tdta0xa405f50) to UDP 127.0.0.1:5060: INVITE sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bKPjM3Mn9WGWMnuOlV0hECET2CdjJO9wwcnj Max-Forwards: 70 From: <sip:192.168.0.2>;tag=b3RASfiSqIt0AK-539TOy5knYThvfBI9 To: sip:127.0.0.1 Contact: <sip:192.168.0.2:5060;ob> Call-ID: 3NhUYiNvyKOQFk-Jm4IP6TYCZIOgyKXw CSeq: 13130 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: 536 v=0 o=- 3552052192 3552052192 IN IP4 192.168.0.2 s=pjmedia c=IN IP4 192.168.0.2 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 192.168.0.2 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 0 RTP/AVP 31 c=IN IP4 127.0.0.1 --end msg-- 18:09:52.345 pjsua_core.c .RX 1143 bytes Request msg INVITE/cseq=13130 (rdata0xa0c86a4) from UDP 127.0.0.1:5060: INVITE sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bKPjM3Mn9WGWMnuOlV0hECET2CdjJO9wwcnj Max-Forwards: 70 From: <sip:192.168.0.2>;tag=b3RASfiSqIt0AK-539TOy5knYThvfBI9 To: sip:127.0.0.1 Contact: <sip:192.168.0.2:5060;ob> Call-ID: 3NhUYiNvyKOQFk-Jm4IP6TYCZIOgyKXw CSeq: 13130 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: 536 v=0 o=- 3552052192 3552052192 IN IP4 192.168.0.2 s=pjmedia c=IN IP4 192.168.0.2 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 192.168.0.2 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 0 RTP/AVP 31 c=IN IP4 127.0.0.1 --end msg-- 18:09:52.345 pjsua_app.c .......Call 0 state changed to CALLING 18:09:52.345 pjsua_call.c !.Incoming Request msg INVITE/cseq=13130 (rdata0xa0c86a4) 18:09:52.346 pjsua_media.c ..Call 1: initializing media.. 18:09:52.346 pjsua_media.c ...RTP socket reachable at 192.168.0.2:40002 18:09:52.346 pjsua_media.c ...RTCP socket reachable at 192.168.0.2:40003 18:09:52.346 pjsua_media.c ...Media index 0 selected for audio call 1 18:09:52.347 pjsua_core.c .....TX 302 bytes Response msg 100/INVITE/cseq=13130 (tdta0xa415790) 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=z9hG4bKPjM3Mn9WGWMnuOlV0hECET2CdjJO9wwcnj Call-ID: 3NhUYiNvyKOQFk-Jm4IP6TYCZIOgyKXw From: <sip:192.168.0.2>;tag=b3RASfiSqIt0AK-539TOy5knYThvfBI9 To: <sip:127.0.0.1> CSeq: 13130 INVITE Content-Length: 0 --end msg-- 18:09:52.347 pjsua_aud.c ..Conf connect: 2 --> 0 18:09:52.347 conference.c ...Port 2 (ring) transmitting to port 0 (Intel 82801DB-ICH4: Intel 82801DB-ICH4 (hw:0,0)) 18:09:52.347 pjsua_app.c ..Incoming call for account 0! Media count: 1 audio & 0 video From: <sip:192.168.0.2> To: <sip:127.0.0.1> Press a to answer or h to reject call 18:09:52.347 pjsua_core.c .RX 302 bytes Response msg 100/INVITE/cseq=13130 (rdata0xa0c86a4) 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=z9hG4bKPjM3Mn9WGWMnuOlV0hECET2CdjJO9wwcnj Call-ID: 3NhUYiNvyKOQFk-Jm4IP6TYCZIOgyKXw From: <sip:192.168.0.2>;tag=b3RASfiSqIt0AK-539TOy5knYThvfBI9 To: <sip:127.0.0.1> CSeq: 13130 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 +=============================================================================+ | 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 | +=============================================================================+ You have 2 active calls Current call id=0 to sip:127.0.0.1 [CALLING] >>> 18:09:52.364 os_core_unix.c Info: possibly re-registering existing thread a No pending incoming call >>> q 18:10:03.453 pjsua_core.c !Shutting down, flags=0... 18:10:03.453 pjsua_core.c PJSUA state changed: RUNNING --> CLOSING ..... rest omitted Provenance : Courrier pour Windows 10 |
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