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トップに関する質問Raspberry Pi:PJSIPより発信、応答した瞬間に切断されてしまう

編集履歴

質問編集履歴

3

Android SIPクライアント変更に伴い構成図を修正

2016/11/19 18:54

投稿

nakky
nakky

スコア10

test CHANGED
File without changes
test CHANGED
@@ -28,7 +28,7 @@
28
28
 
29
29
  ###構成
30
30
 
31
- ![イメージ説明](d3315fb8ba67ecf0dc133ca63d5a894f.jpeg)
31
+ ![イメージ説明](cc5d4255308cac5f5d3e260c6a15e832.jpeg)
32
32
 
33
33
  ###発生している問題・エラーメッセージ
34
34
 

2

asterisk -vvvvr ログ追記

2016/11/19 18:54

投稿

nakky
nakky

スコア10

test CHANGED
File without changes
test CHANGED
File without changes

1

asterisk -vvvvr ログ追記

2016/11/19 01:17

投稿

nakky
nakky

スコア10

test CHANGED
File without changes
test CHANGED
@@ -32,11 +32,49 @@
32
32
 
33
33
  ###発生している問題・エラーメッセージ
34
34
 
35
+ 以下２つのログを添付します。
36
+
37
+ ・asterisk -vvvvr での発信・切断時のログ
38
+
35
- pjsuaより、Make callにて発信・切断のログを添付いたします。
39
+ ・pjsuaより、Make callにて発信・切断時のログ
36
-
40
+
37
- （少々長くなってしまい申し訳ありません。また、ログが長いため、一部削除しております。）
41
+ （ログが長いため、一部削除しております。）
42
+
43
+
44
+
38
-
45
+ [ asterisk ログ ]
46
+
39
-
47
+ ```
48
+
49
+ Connected to Asterisk 1.8.13.1~dfsg1-3+deb7u4 currently running on raspberrypi (pid = 2161)
50
+
51
+ Verbosity was 0 and is now 4
52
+
53
+ -- Registered SIP '32' at 192.168.11.204:5061
54
+
55
+ == Using SIP RTP CoS mark 5
56
+
57
+ -- Executing [30@default:1] Dial("SIP/32-00000004", "SIP/30,30") in new stack
58
+
59
+ == Using SIP RTP CoS mark 5
60
+
61
+ -- Called SIP/30
62
+
63
+ -- SIP/30-00000005 is ringing
64
+
65
+ [Nov 19 09:49:54] NOTICE[2569]: res_rtp_asterisk.c:2339 ast_rtp_read: Unknown RTP codec 95 received from '(null)'
66
+
67
+ -- SIP/30-00000005 answered SIP/32-00000004
68
+
69
+ -- Locally bridging SIP/32-00000004 and SIP/30-00000005
70
+
71
+ == Spawn extension (default, 30, 1) exited non-zero on 'SIP/32-00000004'
72
+
73
+ ```
74
+
75
+
76
+
77
+ [ pjsip ログ ]
40
78
 
41
79
  ```
42
80
 
@@ -256,9 +294,15 @@
256
294
 
257
295
  --end msg--
258
296
 
297
+ 22:41:27.244 pjsua_aud.c .....Conf connect: 1 --> 0
298
+
299
+ 22:41:27.245 conference.c ......Port 1 (ringback) transmitting to port 0 (Master/sound)
300
+
301
+ 22:41:27.245 pjsua_app.c .....Call 0 state changed to EARLY (180 Ringing)
302
+
259
- 22:41:27.238 pjsua_core.c .RX 567 bytes Response msg 180/INVITE/cseq=9624 (rdata0xc3d3b4) from UDP 192.168.11.204:5060:
303
+ 22:41:29.994 pjsua_core.c .RX 895 bytes Response msg 200/INVITE/cseq=9624 (rdata0xc3d3b4) from UDP 192.168.11.204:5060:
260
-
304
+
261
- SIP/2.0 180 Ringing
305
+ SIP/2.0 200 OK
262
306
 
263
307
  Via: SIP/2.0/UDP 192.168.11.204:5061;branch=z9hG4bKPjmM11URHgUQW7GNP2eDt6l4MyOlOiOO0Z;received=192.168.11.204;rport=5061
264
308
 
@@ -280,144 +324,108 @@
280
324
 
281
325
  Contact: <sip:30@192.168.11.204:5060>
282
326
 
327
+ Content-Type: application/sdp
328
+
329
+ Content-Length: 300
330
+
331
+
332
+
333
+ v=0
334
+
335
+ o=root 1629173699 1629173699 IN IP4 192.168.11.204
336
+
337
+ s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4
338
+
339
+ c=IN IP4 192.168.11.204
340
+
341
+ t=0 0
342
+
343
+ m=audio 16602 RTP/AVP 3 0 8 96
344
+
345
+ a=rtpmap:3 GSM/8000
346
+
347
+ a=rtpmap:0 PCMU/8000
348
+
349
+ a=rtpmap:8 PCMA/8000
350
+
351
+ a=rtpmap:96 telephone-event/8000
352
+
353
+ a=fmtp:96 0-16
354
+
355
+ a=ptime:20
356
+
357
+ a=sendrecv
358
+
359
+ --end msg--
360
+
361
+ 22:41:30.008 pjsua_app.c .....Call 0 state changed to CONNECTING
362
+
363
+ 22:41:30.010 pjsua_media.c .....Call 0: updating media..
364
+
365
+ 22:41:30.010 pjsua_aud.c ......Audio channel update..
366
+
367
+ 22:41:30.011 strm0xc517cc .......VAD temporarily disabled
368
+
369
+ 22:41:30.012 strm0xc517cc .......Encoder stream started
370
+
371
+ 22:41:30.012 strm0xc517cc .......Decoder stream started
372
+
373
+ 22:41:30.012 pjsua_media.c ......pjsua_aud_channel_update() failed for call_id 0 media 0: Invalid operation (PJ_EINVALIDOP)
374
+
375
+ 22:41:30.013 pjsua_media.c ......Error updating media call00:0: Invalid operation (PJ_EINVALIDOP)
376
+
377
+ 22:41:30.013 pjsua_call.c .....Unable to create media session: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048]
378
+
379
+ 22:41:30.014 pjsua_core.c ........TX 340 bytes Request msg BYE/cseq=9625 (tdta0xc41048) to UDP 192.168.11.204:5060:
380
+
381
+ BYE sip:30@192.168.11.204:5060 SIP/2.0
382
+
383
+ Via: SIP/2.0/UDP 192.168.11.204:5061;rport;branch=z9hG4bKPjwneXgrEzTyf9OgujNoasSp3AA.3vF9l6
384
+
385
+ Max-Forwards: 70
386
+
387
+ From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p
388
+
389
+ To: sip:30@192.168.11.204;tag=as6fc3ea6b
390
+
391
+ Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO
392
+
393
+ CSeq: 9625 BYE
394
+
395
+ Content-Length: 0
396
+
397
+
398
+
399
+ --end msg--
400
+
401
+ 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
402
+
403
+ 22:41:30.023 pjsua_core.c .RX 483 bytes Response msg 200/BYE/cseq=9625 (rdata0xc3d3b4) from UDP 192.168.11.204:5060:
404
+
405
+ SIP/2.0 200 OK
406
+
407
+ Via: SIP/2.0/UDP 192.168.11.204:5061;branch=z9hG4bKPjwneXgrEzTyf9OgujNoasSp3AA.3vF9l6;received=192.168.11.204;rport=5061
408
+
409
+ From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p
410
+
411
+ To: sip:30@192.168.11.204;tag=as6fc3ea6b
412
+
413
+ Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO
414
+
415
+ CSeq: 9625 BYE
416
+
417
+ Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4
418
+
419
+ Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
420
+
421
+ Supported: replaces, timer
422
+
283
423
  Content-Length: 0
284
424
 
285
425
 
286
426
 
287
427
  --end msg--
288
428
 
289
- 22:41:27.244 pjsua_aud.c .....Conf connect: 1 --> 0
290
-
291
- 22:41:27.245 conference.c ......Port 1 (ringback) transmitting to port 0 (Master/sound)
292
-
293
- 22:41:27.245 pjsua_app.c .....Call 0 state changed to EARLY (180 Ringing)
294
-
295
- 22:41:29.994 pjsua_core.c .RX 895 bytes Response msg 200/INVITE/cseq=9624 (rdata0xc3d3b4) from UDP 192.168.11.204:5060:
296
-
297
- SIP/2.0 200 OK
298
-
299
- Via: SIP/2.0/UDP 192.168.11.204:5061;branch=z9hG4bKPjmM11URHgUQW7GNP2eDt6l4MyOlOiOO0Z;received=192.168.11.204;rport=5061
300
-
301
- From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p
302
-
303
- To: sip:30@192.168.11.204;tag=as6fc3ea6b
304
-
305
- Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO
306
-
307
- CSeq: 9624 INVITE
308
-
309
- Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4
310
-
311
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
312
-
313
- Supported: replaces, timer
314
-
315
- Session-Expires: 1800;refresher=uas
316
-
317
- Contact: <sip:30@192.168.11.204:5060>
318
-
319
- Content-Type: application/sdp
320
-
321
- Content-Length: 300
322
-
323
-
324
-
325
- v=0
326
-
327
- o=root 1629173699 1629173699 IN IP4 192.168.11.204
328
-
329
- s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4
330
-
331
- c=IN IP4 192.168.11.204
332
-
333
- t=0 0
334
-
335
- m=audio 16602 RTP/AVP 3 0 8 96
336
-
337
- a=rtpmap:3 GSM/8000
338
-
339
- a=rtpmap:0 PCMU/8000
340
-
341
- a=rtpmap:8 PCMA/8000
342
-
343
- a=rtpmap:96 telephone-event/8000
344
-
345
- a=fmtp:96 0-16
346
-
347
- a=ptime:20
348
-
349
- a=sendrecv
350
-
351
- --end msg--
352
-
353
- 22:41:30.008 pjsua_app.c .....Call 0 state changed to CONNECTING
354
-
355
- 22:41:30.010 pjsua_media.c .....Call 0: updating media..
356
-
357
- 22:41:30.010 pjsua_aud.c ......Audio channel update..
358
-
359
- 22:41:30.011 strm0xc517cc .......VAD temporarily disabled
360
-
361
- 22:41:30.012 strm0xc517cc .......Encoder stream started
362
-
363
- 22:41:30.012 strm0xc517cc .......Decoder stream started
364
-
365
- 22:41:30.012 pjsua_media.c ......pjsua_aud_channel_update() failed for call_id 0 media 0: Invalid operation (PJ_EINVALIDOP)
366
-
367
- 22:41:30.013 pjsua_media.c ......Error updating media call00:0: Invalid operation (PJ_EINVALIDOP)
368
-
369
- 22:41:30.013 pjsua_call.c .....Unable to create media session: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048]
370
-
371
- 22:41:30.014 pjsua_core.c ........TX 340 bytes Request msg BYE/cseq=9625 (tdta0xc41048) to UDP 192.168.11.204:5060:
372
-
373
- BYE sip:30@192.168.11.204:5060 SIP/2.0
374
-
375
- Via: SIP/2.0/UDP 192.168.11.204:5061;rport;branch=z9hG4bKPjwneXgrEzTyf9OgujNoasSp3AA.3vF9l6
376
-
377
- Max-Forwards: 70
378
-
379
- From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p
380
-
381
- To: sip:30@192.168.11.204;tag=as6fc3ea6b
382
-
383
- Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO
384
-
385
- CSeq: 9625 BYE
386
-
387
- Content-Length: 0
388
-
389
-
390
-
391
- --end msg--
392
-
393
- 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
394
-
395
- 22:41:30.023 pjsua_core.c .RX 483 bytes Response msg 200/BYE/cseq=9625 (rdata0xc3d3b4) from UDP 192.168.11.204:5060:
396
-
397
- SIP/2.0 200 OK
398
-
399
- Via: SIP/2.0/UDP 192.168.11.204:5061;branch=z9hG4bKPjwneXgrEzTyf9OgujNoasSp3AA.3vF9l6;received=192.168.11.204;rport=5061
400
-
401
- From: sip:31@192.168.11.204;tag=dY-GFFdWGeLYZM8Khh-x0nOVQiZ-Qo4p
402
-
403
- To: sip:30@192.168.11.204;tag=as6fc3ea6b
404
-
405
- Call-ID: qJJZH.7BBUsvFaEqHvBb3msd.vTKfoLO
406
-
407
- CSeq: 9625 BYE
408
-
409
- Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u4
410
-
411
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
412
-
413
- Supported: replaces, timer
414
-
415
- Content-Length: 0
416
-
417
-
418
-
419
- --end msg--
420
-
421
429
  22:41:30.025 pjsua_aud.c .....Conf disconnect: 1 -x- 0
422
430
 
423
431
  22:41:30.025 conference.c ......Port 1 (ringback) stop transmitting to port 0 (Master/sound)