Hello, I’m new to PJSIP library so please forgive this smaybe silly question. I’m starting the development of a call center app and consider using pjsip for the moment. Basically the goal is to make a simple server application that connects 2 separate calls while staying “between” the two (one use case would be one call arrives, app calls another phone, then liaise them together while recording in the conf. The app should also be able to redirect one of call if needed). I’m at the simple point of making/receiving calls while playing a little with the lib to understand it. For the moment on my debugging machine it’s working fine (windows machine/sound card up and running), I can receive and make calls without problems. But when deploying on testing server (linux machine/no sound card), PJSUA is sending 488 SIP Code after logging a no active media after sdp neg (logs below). The thing is, I don’t need to play audio on the server (though I should be able to record the conf to a wav file) so do I really need a sound card on the server? I tried using pjsua_set_null_snd_dev() but it’s not affecting anything. Can someone please tell me what are my options here? Do I need to get below pjsua lib, or can I just instruct it to use a conf port without sound device? Thanx for your help. 08:30:46.935 os_core_unix.c !pjlib 2.1 for POSIX initialized 08:30:46.936 sip_endpoint.c .Creating endpoint instance... 08:30:46.937 pjlib .select() I/O Queue created (0x137d110) 08:30:46.937 sip_endpoint.c .Module "mod-msg-print" registered 08:30:46.937 sip_transport. .Transport manager created. 08:30:46.937 pjsua_core.c .PJSUA state changed: NULL --> CREATED 08:30:46.937 pjsua_core.c SIP UDP socket reachable at 192.168.0.72:5060 08:30:46.937 udp0x13716f0 SIP UDP transport started, published address is 192.168.0.72:5060 08:30:46.937 sip_endpoint.c .Module "mod-pjsua-log" registered 08:30:46.937 sip_endpoint.c .Module "mod-tsx-layer" registered 08:30:46.937 sip_endpoint.c .Module "mod-stateful-util" registered 08:30:46.937 sip_endpoint.c .Module "mod-ua" registered 08:30:46.937 sip_endpoint.c .Module "mod-100rel" registered 08:30:46.937 sip_endpoint.c .Module "mod-pjsua" registered 08:30:46.937 sip_endpoint.c .Module "mod-invite" registered 08:30:46.946 pa_dev.c ..PortAudio sound library initialized, status=0 08:30:46.946 pa_dev.c ..PortAudio host api count=2 08:30:46.946 pa_dev.c ..Sound device count=0 08:30:46.947 pjlib ..select() I/O Queue created (0x139fe68) 08:30:46.957 sip_endpoint.c .Module "mod-evsub" registered 08:30:46.957 sip_endpoint.c .Module "mod-presence" registered 08:30:46.957 sip_endpoint.c .Module "mod-mwi" registered 08:30:46.957 sip_endpoint.c .Module "mod-refer" registered 08:30:46.957 sip_endpoint.c .Module "mod-pjsua-pres" registered 08:30:46.957 sip_endpoint.c .Module "mod-pjsua-im" registered 08:30:46.957 sip_endpoint.c .Module "mod-pjsua-options" registered 08:30:46.957 pjsua_core.c .1 SIP worker threads created 08:30:46.957 pjsua_core.c .pjsua version 2.1 for Linux-4.4.0.62/x86_64/glibc-2.17 initialized 08:30:46.957 pjsua_core.c .PJSUA state changed: CREATED --> INIT 08:30:46.957 pjsua_aud.c Setting null sound device.. 08:30:46.957 pjsua_aud.c .Opening null sound device.. 08:30:46.957 pjsua_core.c PJSUA state changed: INIT --> STARTING 08:30:46.957 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 08:30:46.957 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING 08:30:46.957 pjsua_acc.c Adding account: id=sip:2000@xxxxxxxxxxxxxxxxxxxxx 08:30:46.958 pjsua_acc.c .Account sip:2000@xxxxxxxxxxxxxxxxxxxxx added with id 0 08:30:46.958 pjsua_acc.c Acc 0: setting registration.. 08:30:46.959 pjsua_core.c ..TX 521 bytes Request msg REGISTER/cseq=10512 (tdta0x13be650) to UDP 192.168.0.71:5060: REGISTER sip:frvln001.sodiware.lan SIP/2.0 Via: SIP/2.0/UDP 192.168.0.72:5060;rport;branch=z9hG4bKPj2c04dbf3-df29-447f-8fca-f71170074040 Max-Forwards: 70 From: <sip:2000@xxxxxxxxxxxxxxxxxxxxx>;tag=ff67eaac-8b15-4757-b998-0465ba37aa45 To: <sip:2000@xxxxxxxxxxxxxxxxxxxxx> Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487 CSeq: 10512 REGISTER Contact: <sip:2000@192.168.0.72:5060;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- 08:30:46.959 pjsua_acc.c .Acc 0: Registration sent 08:30:46.960 pjsua_core.c .RX 608 bytes Response msg 401/REGISTER/cseq=10512 (rdata0x13867a8) from UDP 192.168.0.71:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.72:5060;branch=z9hG4bKPj2c04dbf3-df29-447f-8fca-f71170074040;received=192.168.0.72;rport=5060 From: <sip:2000@xxxxxxxxxxxxxxxxxxxxx>;tag=ff67eaac-8b15-4757-b998-0465ba37aa45 To: <sip:2000@xxxxxxxxxxxxxxxxxxxxx>;tag=as291f3de5 Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487 CSeq: 10512 REGISTER Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sodiware.lan", nonce="0562418a" Content-Length: 0 --end msg-- 08:30:46.960 pjsua_core.c ....TX 693 bytes Request msg REGISTER/cseq=10513 (tdta0x13be650) to UDP 192.168.0.71:5060: REGISTER sip:frvln001.sodiware.lan SIP/2.0 Via: SIP/2.0/UDP 192.168.0.72:5060;rport;branch=z9hG4bKPj98d7f725-60d5-4af2-91a0-9a8d96bc59a4 Max-Forwards: 70 From: <sip:2000@xxxxxxxxxxxxxxxxxxxxx>;tag=ff67eaac-8b15-4757-b998-0465ba37aa45 To: <sip:2000@xxxxxxxxxxxxxxxxxxxxx> Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487 CSeq: 10513 REGISTER Contact: <sip:2000@192.168.0.72:5060;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="2000", realm="sodiware.lan", nonce="0562418a", uri="sip:frvln001.sodiware.lan", response="ca2dd328bfa588ca93b23599d3f71551", algorithm=MD5 Content-Length: 0 --end msg-- 08:30:46.961 pjsua_core.c .RX 623 bytes Response msg 200/REGISTER/cseq=10513 (rdata0x13867a8) from UDP 192.168.0.71:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.72:5060;branch=z9hG4bKPj98d7f725-60d5-4af2-91a0-9a8d96bc59a4;received=192.168.0.72;rport=5060 From: <sip:2000@xxxxxxxxxxxxxxxxxxxxx>;tag=ff67eaac-8b15-4757-b998-0465ba37aa45 To: <sip:2000@xxxxxxxxxxxxxxxxxxxxx>;tag=as291f3de5 Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487 CSeq: 10513 REGISTER Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 300 Contact: <sip:2000@192.168.0.72:5060;ob>;expires=300 Date: Sat, 11 Feb 2017 01:57:47 GMT Content-Length: 0 --end msg-- 08:30:46.961 pjsua_acc.c ....SIP outbound status for acc 0 is not active 08:30:46.961 pjsua_acc.c ....sip:2000@xxxxxxxxxxxxxxxxxxxxx: registration success, status=200 (OK), will re-register in 300 seconds 08:30:46.961 pjsua_acc.c ....Keep-alive timer started for acc 0, destination:192.168.0.71:5060, interval:15s 08:31:04.791 pjsua_core.c .RX 882 bytes Request msg INVITE/cseq=102 (rdata0x13867a8) from UDP 192.168.0.71:5060: INVITE sip:2000@192.168.0.72:5060;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.0.71:5060;branch=z9hG4bK751edb1f Max-Forwards: 70 From: "New User" <sip:5000@192.168.0.71>;tag=as1e417004 To: <sip:2000@192.168.0.72:5060;ob> Contact: <sip:5000@192.168.0.71:5060> Call-ID: 09302cad1054f32764b853443303030b@192.168.0.71:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3 Date: Sat, 11 Feb 2017 01:58:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 296 v=0 o=root 595539155 595539155 IN IP4 192.168.0.71 s=Asterisk PBX 1.8.13.1~dfsg-3ubuntu3 c=IN IP4 192.168.0.71 t=0 0 m=audio 10616 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --end msg-- 08:31:04.791 pjsua_call.c .Incoming Request msg INVITE/cseq=102 (rdata0x13867a8) 08:31:04.791 pjsua_media.c ..Call 0: initializing media.. 08:31:04.791 pjsua_media.c ...RTP socket reachable at 192.168.0.72:40000 08:31:04.791 pjsua_media.c ...RTCP socket reachable at 192.168.0.72:40001 08:31:04.791 pjsua_media.c ...Media index 0 selected for audio call 0 08:31:04.792 /home/SODIWARE ..>>>>> Remote session from sdp form <<<< 08:31:04.792 pjsua_core.c .....TX 291 bytes Response msg 100/INVITE/cseq=102 (tdta0x13ccc50) to UDP 192.168.0.71:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.71:5060;received=192.168.0.71;branch=z9hG4bK751edb1f Call-ID: 09302cad1054f32764b853443303030b@192.168.0.71:5060 From: "New User" <sip:5000@192.168.0.71>;tag=as1e417004 To: <sip:2000@192.168.0.72;ob> CSeq: 102 INVITE Content-Length: 0 --end msg-- 08:31:04.792 /home/SODIWARE ..Waiting 2 seconds : 1 08:31:06.792 pjsua_call.c ..Answering call 0: code=200 08:31:06.792 pjsua_media.c .....Call 0: updating media.. 08:31:06.792 pjsua_aud.c ......Audio channel update.. 08:31:06.793 strm0x13d1a98 .......VAD temporarily disabled 08:31:06.793 strm0x13d1a98 .......Encoder stream started 08:31:06.793 strm0x13d1a98 .......Decoder stream started 08:31:06.793 pjsua_media.c ......pjsua_aud_channel_update() failed ftitanslaved.out: /home/SODIWARE/kpamafre/projects/swsiprt/IncomingCall.cpp:31: virtual HRESULT CIncomingCall::Answer(): Assertion `status == PJ_SUCCESS' failed. or call_id 0 media 0: Invalid operation (PJ_EINVALIDOP) 08:31:06.793 pjsua_call.c .....Unable to create media session: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048] 08:31:06.793 pjsua_core.c ........TX 491 bytes Response msg 488/INVITE/cseq=102 (tdta0x13ccc50) to UDP 192.168.0.71:5060: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.0.71:5060;received=192.168.0.71;branch=z9hG4bK751edb1f Call-ID: 09302cad1054f32764b853443303030b@192.168.0.71:5060 From: "New User" <sip:5000@192.168.0.71>;tag=as1e417004 To: <sip:2000@192.168.0.72;ob>;tag=baa5cf52-af70-42a0-a919-5d511bfcfb78 CSeq: 102 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Length: 0 --end msg-- 08:31:06.793 /home/SODIWARE ...........Found agent 0 for call -781117144 08:31:06.793 /home/SODIWARE ...........Call state changed (id: -781115088) 08:31:06.793 pjsua_media.c ...........Call 0: deinitializing media.. 08:31:06.793 pjsua_media.c .............Media stream call00:0 is destroyed 08:31:06.793 pjsua_call.c ...Error creating response: Not Acceptable Here [status=170488] |
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