Webrtc with asterisk behind NAT


#1

Hi
I’m using webrtc to register extension on asterisk 14 running behind NAT, external_ip localnet and other need settings are done. But call drops after few hours with hangupcause code 44. I checked settings on asterisk, I also run stun and turn server behind NAT. Anyone can help?


#2

You’d need to provide the console output and “pjsip set logger on” output to show exactly who and what is hanging up the call.


#3

Now it’s change to cause 16, this is the output, i’m not using pjsip but chan_sip for webrtc client.

<------------>
[2017-11-08 17:39:03] VERBOSE[29072] chan_sip.c: Scheduling destruction of SIP dialog ‘e06kojq6u9qa10bv66uj2o’ in 32000 ms (Method: REGISTER)
[2017-11-08 17:39:19] VERBOSE[24738] chan_sip.c: Really destroying SIP dialog ‘3mv3uh5lkgoibj0uhsul7h’ Method: REGISTER
[2017-11-08 17:39:19] VERBOSE[24198] chan_sip.c:
<— SIP read from WS:114.236.90.213:54774 —>
BYE sip:6019@218.4.144.4:0;transport=ws SIP/2.0
Via: SIP/2.0/WSS s5pa2pn3ni79.invalid;branch=z9hG4bK5106063
Max-Forwards: 69
To: sip:6019@218.4.144.4:0;tag=as5bc78167
From: “6001” sip:08133b3e@s5pa2pn3ni79.invalid;transport=ws;tag=csd7vtn0qi
Call-ID: 6c6d7b9a00a5e99e14c3a2ba4fccc21a@218.4.144.4:0
CSeq: 1494 BYE
Reason: SIP ;cause=408; text="RTP Timeout"
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.0.15
Content-Length: 0

<------------->
[2017-11-08 17:39:19] VERBOSE[24198] chan_sip.c: — (12 headers 0 lines) —
[2017-11-08 17:39:19] VERBOSE[24198][C-0000005c] chan_sip.c: Scheduling destruction of SIP dialog ‘6c6d7b9a00a5e99e14c3a2ba4fccc21a@218.4.144.4:0’ in 32000 ms (Method: BYE)
[2017-11-08 17:39:19] VERBOSE[24198][C-0000005c] chan_sip.c:
<— Transmitting (no NAT) to 114.236.90.213:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS s5pa2pn3ni79.invalid;branch=z9hG4bK5106063;received=114.236.90.213
From: “6001” sip:08133b3e@s5pa2pn3ni79.invalid;transport=ws;tag=csd7vtn0qi
To: sip:6019@218.4.144.4:0;tag=as5bc78167
Call-ID: 6c6d7b9a00a5e99e14c3a2ba4fccc21a@218.4.144.4:0
CSeq: 1494 BYE
Server: FPBX-14.0.1.1(14.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2017-11-08 17:39:19] VERBOSE[24295][C-0000005c] bridge_channel.c: Channel SIP/6001-000000ab left ‘simple_bridge’ basic-bridge
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] bridge_channel.c: Channel SIP/6019-000000aa left ‘simple_bridge’ basic-bridge
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] app_macro.c: Spawn extension (macro-dial-one, s, 53) exited non-zero on ‘SIP/6019-000000aa’ in macro ‘dial-one’
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] app_macro.c: Spawn extension (macro-exten-vm, s, 20) exited non-zero on ‘SIP/6019-000000aa’ in macro ‘exten-vm’
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] pbx.c: Spawn extension (from-internal, 6001, 2) exited non-zero on ‘SIP/6019-000000aa’
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] pbx.c: Executing [h@from-internal:1] Macro(“SIP/6019-000000aa”, “hangupcall”) in new stack
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/6019-000000aa”, “1?theend”) in new stack
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/6019-000000aa”, “0?Set(CDR(recordingfile)=)”) in new stack
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“SIP/6019-000000aa”, “”) in new stack
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/6019-000000aa’ in macro ‘hangupcall’
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/6019-000000aa’
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] chan_sip.c: Scheduling destruction of SIP dialog ‘733blf5lngrbj3sgvv43’ in 32000 ms (Method: ACK)
[2017-11-08 17:39:19] VERBOSE[24294][C-0000005c] chan_sip.c: Reliably Transmitting (NAT) to 192.9.200.158:33870:
BYE sip:c1psti7o@ijiju8opt50n.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.60.128:5160;branch=z9hG4bK54bd01fa;rport
Max-Forwards: 70
From: sip:6001@192.168.60.128;tag=as4b606b0e
To: “6019” sip:6019@192.168.60.128;tag=lbohf3vbeb
Call-ID: 733blf5lngrbj3sgvv43
CSeq: 102 BYE
User-Agent: FPBX-14.0.1.1(14.5.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


[2017-11-08 17:39:19] VERBOSE[29072] chan_sip.c:
<— SIP read from WS:192.9.200.158:33870 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.60.128:5160;branch=z9hG4bK54bd01fa;rport
To: “6019” sip:6019@192.168.60.128;tag=lbohf3vbeb
From: sip:6001@192.168.60.128;tag=as4b606b0e
Call-ID: 733blf5lngrbj3sgvv43
CSeq: 102 BYE
Supported: outbound
Content-Length: 0

<------------->
[2017-11-08 17:39:19] VERBOSE[29072] chan_sip.c: — (8 headers 0 lines) —
[2017-11-08 17:39:19] VERBOSE[29072][C-0000005c] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2017-11-08 17:39:20] VERBOSE[24738] chan_sip.c: Really destroying SIP dialog ‘733blf5lngrbj3sgvv43’ Method: ACK
[2017-11-08 17:39:27] VERBOSE[2063] chan_sip.c:
<— SIP read from WS:192.9.200.224:61292 —>
REGISTER sip:192.168.60.128 SIP/2.0
Via: SIP/2.0/WSS j2o2c8176qjs.invalid;branch=z9hG4bK1049798
Max-Forwards: 69
To: sip:6014@192.168.60.128
From: “6014” sip:6014@192.168.60.128;tag=hufpb2pebk
Call-ID: lgfk3tfcoh01k49runor8s
CSeq: 633 REGISTER
Contact: sip:gcv73i18@j2o2c8176qjs.invalid;transport=ws;+sip.ice;reg-id=1;+sip.instance=“urn:uuid:127b6fed-2075-4dfe-855b-21418330a3aa”;expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 3.0.15
Content-Length: 0

<------------->
[2017-11-08 17:39:27] VERBOSE[2063] chan_sip.c: — (13 headers 0 lines) —
[2017-11-08 17:39:27] VERBOSE[2063] chan_sip.c:
<— Transmitting (no NAT) to 192.9.200.224:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS j2o2c8176qjs.invalid;branch=z9hG4bK1049798;received=192.9.200.224
From: “6014” sip:6014@192.168.60.128;tag=hufpb2pebk
To: sip:6014@192.168.60.128;tag=as65d5cd62
Call-ID: lgfk3tfcoh01k49runor8s
CSeq: 633 REGISTER
Server: FPBX-14.0.1.1(14.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="78c1d46b"
Content-Length: 0