[asterisk-bugs] [JIRA] (ASTERISK-27381) Crash inside opus codec
Torrey Searle (JIRA)
noreply at issues.asterisk.org
Fri Nov 17 03:01:41 CST 2017
[ https://issues.asterisk.org/jira/browse/ASTERISK-27381?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=240097#comment-240097 ]
Torrey Searle commented on ASTERISK-27381:
------------------------------------------
The test call I was doing passes through asterisk twice, I filtered the log on the channel id so you only have the channel with the issue in this log
{noformat}
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] pbx.c: Launching 'AGI'
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] pbx.c: Executing [3227470203 at default:1] AGI("PJSIP/srnd0ser-000000a2", "agi://127.0.0.1:20203/agi_load_balance_handler") in new stack
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] res_agi.c: Wow, connected!
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] res_agi.c: agi://127.0.0.1:20203/agi_load_balance_handler: [ast13sip01] request for VCC node received
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] res_agi.c: agi://127.0.0.1:20203/agi_load_balance_handler: statement : SELECT ms.ip, ms.max_calls FROM callcontrollocal.machine_status ms WHERE ms.hostname LIKE ? AND ms.service = vcc AND IFNULL(ms.force_alive, ms.is_alive) = 1;
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] res_agi.c: agi://127.0.0.1:20203/agi_load_balance_handler: selected VCC node (in pop ast13) : 10.1.25.3
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] res_agi.c: <PJSIP/srnd0ser-000000a2>AGI Script agi://127.0.0.1:20203/agi_load_balance_handler completed, returning 0
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] pbx.c: Launching 'NoOp'
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] pbx.c: Executing [3227470203 at default:2] NoOp("PJSIP/srnd0ser-000000a2", "using VCC node : 10.1.25.3") in new stack
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] pbx_variables.c: Function PJSIP_HEADER(read,X-VCC-CallType) result is '(null)'
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] pbx_variables.c: Expression result is '0'
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] pbx.c: Launching 'GotoIf'
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] pbx.c: Executing [3227470203 at default:3] GotoIf("PJSIP/srnd0ser-000000a2", "0?voxfax-init,,1:") in new stack
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] pbx_builtins.c: Not taking any branch
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] pbx.c: Launching 'AGI'
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] pbx.c: Executing [3227470203 at default:4] AGI("PJSIP/srnd0ser-000000a2", "agi://10.1.25.3/init") in new stack
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] res_agi.c: Wow, connected!
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] res_agi.c: <PJSIP/srnd0ser-000000a2>AGI Script agi://10.1.25.3/init completed, returning 0
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] pbx.c: Launching 'Answer'
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] pbx.c: Executing [echo at ivrs2:1] Answer("PJSIP/srnd0ser-000000a2", "") in new stack
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] res_rtp_asterisk.c: 0x7f3f781151f0 -- Probation learning mode pass with source address 10.1.25.2:12238
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] chan_pjsip.c: Oooh, got a frame with format of opus on channel 'PJSIP/srnd0ser-000000a2' when we're sending 'alaw', switching to match
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] channel.c: Channel PJSIP/srnd0ser-000000a2 setting write format path: alaw -> opus
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] channel.c: Channel PJSIP/srnd0ser-000000a2 setting read format path: alaw -> opus
[Nov 17 08:52:55] DEBUG[2430][C-0000009c] pbx.c: Launching 'Wait'
[Nov 17 08:52:55] VERBOSE[2430][C-0000009c] pbx.c: Executing [echo at ivrs2:2] Wait("PJSIP/srnd0ser-000000a2", "4") in new stack
[Nov 17 08:52:59] DEBUG[2430][C-0000009c] pbx.c: Launching 'Playback'
[Nov 17 08:52:59] VERBOSE[2430][C-0000009c] pbx.c: Executing [echo at ivrs2:3] Playback("PJSIP/srnd0ser-000000a2", "demo-echotest") in new stack
[Nov 17 08:52:59] DEBUG[2430][C-0000009c] channel.c: Channel PJSIP/srnd0ser-000000a2 setting write format path: gsm -> opus
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] DEBUG[2430][C-0000009c] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Nov 17 08:52:59] VERBOSE[2430][C-0000009c] file.c: <PJSIP/srnd0ser-000000a2> Playing 'demo-echotest.gsm' (language 'en')
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
[Nov 17 08:52:59] ERROR[2430][C-0000009c] codec_opus.c: Opus: failed to create encoder: invalid argument
{noformat}
> Crash inside opus codec
> -----------------------
>
> Key: ASTERISK-27381
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-27381
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Codecs/codec_opus
> Affects Versions: 13.15.1
> Reporter: Torrey Searle
> Assignee: Torrey Searle
> Attachments: codecs.conf, core.asterisk.ast13sip01-1509443936-11-brief.txt, core.asterisk.ast13sip01-1509443936-11-full.txt, core.asterisk.ast13sip01-1509443936-11-locks.txt, core.asterisk.ast13sip01-1509443936-11-thread1.txt
>
>
> Crash has been detected on our WEBRTC platform inside the opus codec version 1.1.0 attached is the backtrace
> {noformat}
> #0 0x00007f0102b5c642 in ?? () from /usr/lib/asterisk/modules/codec_opus-13.0_1.1.0-x86_64.so
> #1 0x00007f0102b50f26 in ?? () from /usr/lib/asterisk/modules/codec_opus-13.0_1.1.0-x86_64.so
> #2 0x00000000005dd38e in framein (f=<optimized out>, pvt=<optimized out>) at translate.c:423
> #3 ast_translate (path=0x7f00ec02cdd0, f=0x7f0030004560, consume=0) at translate.c:573
> #4 0x00000000004b9c3c in ast_write (chan=0x7f008837fec0, fr=0x7f0030004560) at channel.c:5290
> #5 0x00000000004822ea in bridge_channel_handle_write (bridge_channel=<optimized out>) at bridge_channel.c:2346
> #6 bridge_channel_wait (bridge_channel=<optimized out>) at bridge_channel.c:2593
> #7 bridge_channel_internal_join (bridge_channel=0x7f00ec0080b0) at bridge_channel.c:2728
> #8 0x000000000046c3d6 in ast_bridge_join (bridge=bridge at entry=0x7f00ec00d120, chan=chan at entry=0x7f008837fec0, swap=swap at entry=0x0, features=features at entry=0x7f00aab08a10,
> tech_args=tech_args at entry=0x0, flags=flags at entry=(AST_BRIDGE_JOIN_PASS_REFERENCE | AST_BRIDGE_JOIN_INHIBIT_JOIN_COLP)) at bridge.c:1713
> #9 0x000000000050536f in ast_bridge_call_with_flags (chan=chan at entry=0x7f008837fec0, peer=peer at entry=0x7f00ec015bf0, config=config at entry=0x7f00aab08e00, flags=flags at entry=0)
> at features.c:672
> #10 0x0000000000505477 in ast_bridge_call (chan=chan at entry=0x7f008837fec0, peer=peer at entry=0x7f00ec015bf0, config=config at entry=0x7f00aab08e00) at features.c:711
> #11 0x00007f0056643019 in dial_exec_full (chan=0x7f008837fec0, data=<optimized out>, peerflags=peerflags at entry=0x7f00aab09740, continue_exec=continue_exec at entry=0x0)
> at app_dial.c:3224
> #12 0x00007f0056644126 in dial_exec (chan=<optimized out>, data=<optimized out>) at app_dial.c:3280
> #13 0x0000000000579d5e in pbx_exec (c=c at entry=0x7f008837fec0, app=app at entry=0x2224550, data=data at entry=0x7f00aab09c50 "PJSIP/883510080318 at cnhk1ser,180,b(predial^s^1)")
> at pbx_app.c:491
> #14 0x000000000056ee41 in pbx_extension_helper (c=0x7f008837fec0, context=0x7f0088380878 "webrtc", exten=0x7f00883808c8 "s", priority=1, label=<optimized out>,
> callerid=<optimized out>, action=E_SPAWN, found=0x7f00aab0bcec, combined_find_spawn=1, con=0x0) at pbx.c:2884
> #15 0x0000000000570fd9 in ast_spawn_extension (combined_find_spawn=<optimized out>, found=<optimized out>, callerid=<optimized out>, priority=<optimized out>,
> exten=<optimized out>, context=<optimized out>, c=<optimized out>) at pbx.c:4109
> #16 __ast_pbx_run (c=0x7f008837fec0, args=0x7f00ec02cec0, args at entry=0x0) at pbx.c:4286
> #17 0x000000000057251b in pbx_thread (data=data at entry=0x7f008837fec0) at pbx.c:4608
> #18 0x00000000005e2eda in dummy_start (data=<optimized out>) at utils.c:1238
> #19 0x00007f0115bb1064 in start_thread (arg=0x7f00aab0c700) at pthread_create.c:309
> {noformat}
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