###前提・実現したいこと
PJSIPにてAsteriskと接続、他のSIPクライアントとの通信を行いたい。
Make Call でSIPクライアントへ発信できるが、
SIPクライアントで応答した瞬間に、切断されてしまう。
解決策を教えていただければと思います。
【補足】
2台のAndroid端末にインストールした SIPクライアント間で通話は確認済み
上記のSIPクライアントへPJSIPから発信した場合にのみ事象が発生
Asteriskをインストールしているハード/OS は以下となります。
Raspberry Pi 2
Linux raspberrypi 4.1.19-v7+ #858 SMP Tue Mar 15 15:56:00 GMT 2016 armv7l GNU/Linux
###構成
###発生している問題・エラーメッセージ
以下2つのログを添付します。
・asterisk -vvvvr での発信・切断時のログ
・pjsuaより、Make callにて発信・切断時のログ
(ログが長いため、一部削除しております。)
[ asterisk ログ ]
Connected to Asterisk 1.8.13.1~dfsg1-3+deb7u4 currently running on raspberrypi (pid = 2161) Verbosity was 0 and is now 4 -- Registered SIP '32' at 192.168.11.204:5061 == Using SIP RTP CoS mark 5 -- Executing [30@default:1] Dial("SIP/32-00000004", "SIP/30,30") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/30 -- SIP/30-00000005 is ringing [Nov 19 09:49:54] NOTICE[2569]: res_rtp_asterisk.c:2339 ast_rtp_read: Unknown RTP codec 95 received from '(null)' -- SIP/30-00000005 answered SIP/32-00000004 -- Locally bridging SIP/32-00000004 and SIP/30-00000005 == Spawn extension (default, 30, 1) exited non-zero on 'SIP/32-00000004'
[ pjsip ログ ]
Make call: sip:30@192.168.11.204 22:41:26.717 pjsua_call.c !Making call with acc #2 to sip:30@192.168.11.204 22:41:26.717 pjsua_aud.c .Set sound device: capture=-99, playback=-99 22:41:26.717 pjsua_aud.c ..Setting null sound device.. 22:41:26.717 pjsua_app.c ...Turning sound device ON 22:41:26.717 pjsua_aud.c ...Opening null sound device.. 22:41:26.718 pjsua_media.c .Call 0: initializing media.. 22:41:26.719 pjsua_media.c ..RTP socket reachable at 192.168.11.204:4000 22:41:26.719 pjsua_media.c ..RTCP socket reachable at 192.168.11.204:4001 22:41:26.719 pjsua_media.c ..Media index 0 selected for audio call 0 22:41:26.722 pjsua_core.c ....TX 1119 bytes Request msg INVITE/cseq=9623 (tdta0xc48218) to UDP 192.168.11.204:5060: INVITE sip:30@192.168.11.204 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.204:5061;rport;branch=z9hG4bKPjGRVw8t3Z.xhPtTKMcXwT-sS1nPDGtmFG Max-Forwards: 70 From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p To: sip:30@192.168.11.204 Contact: <sip:31@192.168.11.204:5061;ob> Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO CSeq: 9623 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 v2.5.5 Linux-4.1.19/armv7l/glibc-2.13 Content-Type: application/sdp Content-Length: 479 v=0 o=- 3688206086 3688206086 IN IP4 192.168.11.204 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.11.204 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.11.204 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 0-16 --end msg-- 22:41:26.726 pjsua_app.c .......Call 0 state changed to CALLING >>> 22:41:26.728 pjsua_core.c .RX 572 bytes Response msg 401/INVITE/cseq=9623 (rdata0xc3d3b4) from UDP 192.168.11.204:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.11.204:5061;branch=z9hG4bKPjGRVw8t3Z.xhPtTKMcXwT-sS1nPDGtmFG;received=192.168.11.204;rport=5061 From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p To: sip:30@192.168.11.204;tag=as78823107 Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO CSeq: 9623 INVITE Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6230a0dc" Content-Length: 0 --end msg-- 22:41:26.739 pjsua_core.c .......TX 1281 bytes Request msg INVITE/cseq=9624 (tdta0xc48218) to UDP 192.168.11.204:5060: INVITE sip:30@192.168.11.204 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.204:5061;rport;branch=z9hG4bKPjmM11URHgUQW7GNP2eDt6l4MyOlOiOO0Z Max-Forwards: 70 From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p To: sip:30@192.168.11.204 Contact: <sip:31@192.168.11.204:5061;ob> Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO CSeq: 9624 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 v2.5.5 Linux-4.1.19/armv7l/glibc-2.13 Authorization: Digest username="31", realm="asterisk", nonce="6230a0dc", uri="sip:30@192.168.11.204", response="25d92327951bc9291b1732b2cfdfeae0", algorithm=MD5 Content-Type: application/sdp Content-Length: 479 v=0 o=- 3688206086 3688206086 IN IP4 192.168.11.204 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.11.204 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.11.204 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 0-16 --end msg-- 22:41:27.244 pjsua_aud.c .....Conf connect: 1 --> 0 22:41:27.245 conference.c ......Port 1 (ringback) transmitting to port 0 (Master/sound) 22:41:27.245 pjsua_app.c .....Call 0 state changed to EARLY (180 Ringing) 22:41:29.994 pjsua_core.c .RX 895 bytes Response msg 200/INVITE/cseq=9624 (rdata0xc3d3b4) from UDP 192.168.11.204:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.204:5061;branch=z9hG4bKPjmM11URHgUQW7GNP2eDt6l4MyOlOiOO0Z;received=192.168.11.204;rport=5061 From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p To: sip:30@192.168.11.204;tag=as6fc3ea6b Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO CSeq: 9624 INVITE Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:30@192.168.11.204:5060> Content-Type: application/sdp Content-Length: 300 v=0 o=root 1629173699 1629173699 IN IP4 192.168.11.204 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 c=IN IP4 192.168.11.204 t=0 0 m=audio 16602 RTP/AVP 3 0 8 96 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=sendrecv --end msg-- 22:41:30.008 pjsua_app.c .....Call 0 state changed to CONNECTING 22:41:30.010 pjsua_media.c .....Call 0: updating media.. 22:41:30.010 pjsua_aud.c ......Audio channel update.. 22:41:30.011 strm0xc517cc .......VAD temporarily disabled 22:41:30.012 strm0xc517cc .......Encoder stream started 22:41:30.012 strm0xc517cc .......Decoder stream started 22:41:30.012 pjsua_media.c ......pjsua_aud_channel_update() failed for call_id 0 media 0: Invalid operation (PJ_EINVALIDOP) 22:41:30.013 pjsua_media.c ......Error updating media call00:0: Invalid operation (PJ_EINVALIDOP) 22:41:30.013 pjsua_call.c .....Unable to create media session: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048] 22:41:30.014 pjsua_core.c ........TX 340 bytes Request msg BYE/cseq=9625 (tdta0xc41048) to UDP 192.168.11.204:5060: BYE sip:30@192.168.11.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.204:5061;rport;branch=z9hG4bKPjwneXgrEzTyf9OgujNoasSp3AA.3vF9l6 Max-Forwards: 70 From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p To: sip:30@192.168.11.204;tag=as6fc3ea6b Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO CSeq: 9625 BYE Content-Length: 0 --end msg-- 22:41:30.022 sip_endpoint.c .Response msg 487/INVITE/cseq=9624 (rdata0xc3d3b4) from 192.168.11.204:5060 was dropped/unhandled by any modules 22:41:30.023 pjsua_core.c .RX 483 bytes Response msg 200/BYE/cseq=9625 (rdata0xc3d3b4) from UDP 192.168.11.204:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.204:5061;branch=z9hG4bKPjwneXgrEzTyf9OgujNoasSp3AA.3vF9l6;received=192.168.11.204;rport=5061 From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p To: sip:30@192.168.11.204;tag=as6fc3ea6b Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO CSeq: 9625 BYE Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --end msg-- 22:41:30.025 pjsua_aud.c .....Conf disconnect: 1 -x- 0 22:41:30.025 conference.c ......Port 1 (ringback) stop transmitting to port 0 (Master/sound) 22:41:30.025 pjsua_app.c .....Call 0 is DISCONNECTED [reason=488 (Not Acceptable Here)] 22:41:30.026 pjsua_app_comm ..... [DISCONNCTD] To: sip:30@192.168.11.204;tag=as6fc3ea6b Call time: 00h:00m:00s, 1st res in 526 ms, conn in 3300ms #0 audio deactivated 22:41:30.026 pjsua_media.c .....Call 0: deinitializing media.. Segmentation fault