Disconnecting channel for lack of RTP activity in 60 seconds

progressaurus · June 18, 2020, 4:24pm

Hello,

I can successfully make a call using sipML5 from a chrome browser to an external mobile phone. Connection is fine with no issue with audio either. Problem is when I hang up from the external mobile device the sipML5 phone does not hang up. I’ve put debugging messages into the sipML5 code however there seems to be no event recieved when the external device is hung up.

Instead the I recieve the error:

Disconnecting channel 'PJSIP/carrierout-0000001b' for lack of RTP activity in 60 seconds

The call then hangs up and I see the expected messages to deal with the call termination. If I hangup from the sipML5 device the call hangs up as expected and termiates the call on the external mobile device too.

Has anyone else experienced this?

Thanks for any help.

I am using centos7 and asterisk 17.3.0 the build was --with-jansson-bundled

sipML5 Log:

State machine: c0000_Started_2_Outgoing_X_oINVITE
tsk_utils.js?svn=252:115 ICE servers:[{"url":"stun:stun.l.google.com:19302"},{"url":"stun:stun.counterpath.net:3478"},{"url":"stun:numb.viagenie.ca:3478"}]
tsk_utils.js?svn=252:115 ==stack event = m_permission_requested
tsk_utils.js?svn=252:115 ==session event = connecting
tsk_utils.js?svn=252:115 onGetUserMediaSuccess
tsk_utils.js?svn=252:115 createOffer
tsk_utils.js?svn=252:115 onNegotiationNeeded
tsk_utils.js?svn=252:115 onCreateSdpSuccess
tsk_utils.js?svn=252:115 ==stack event = m_permission_accepted
tsk_utils.js?svn=252:115 onSignalingstateChange:have-local-offer
tsk_utils.js?svn=252:115 onSetLocalDescriptionSuccess
10tsk_utils.js?svn=252:115 onIceCandidate = gathering
tsk_utils.js?svn=252:115 onIceCandidate = complete
tsk_utils.js?svn=252:115 ICE GATHERING COMPLETED!
tsk_utils.js?svn=252:115 onIceGatheringCompleted
tsk_utils.js?svn=252:115 SEND: INVITE sip:077XXXXXXXX@NULL SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKNUr3Doml8vlIgTKPuTCkLRhSgTUfAThb;rport
From: "3000"<sip:3000@192.168.1.7>;tag=KIVWD4blxGuIakHafyEu
To: <sip:077XXXXXXXX@NULL>
Contact: "3000"<sips:3000@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;impi=3000;ha1=b57e29d9ee165b9ee6a9457a734b8cac;+g.oma.sip-im;language="en,fr"
Call-ID: 342413b2-30db-06d8-6b18-fcc16be87a7d
CSeq: 17323 INVITE
Content-Type: application/sdp
Content-Length: 2821
Max-Forwards: 70
User-Agent: 199
Organisation: callcent

v=0
o=- 2062517073951919000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS R7XC1bG60xk5SCT9NcJHV63VrWekNgUfgKnm
m=audio 35116 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 XX.XXX.XX.XX
a=rtcp:35118 IN IP4 XX.XXX.XX.XX
a=candidate:1230308205 1 udp 2122260223 192.168.1.62 57708 typ host generation 0 network-id 2
a=candidate:250968509 1 udp 2122194687 192.168.1.52 57709 typ host generation 0 network-id 1 network-cost 10
a=candidate:1230308205 2 udp 2122260222 192.168.1.62 57710 typ host generation 0 network-id 2
a=candidate:250968509 2 udp 2122194686 192.168.1.52 57711 typ host generation 0 network-id 1 network-cost 10
a=candidate:131530653 1 tcp 1518280447 192.168.1.62 9 typ host tcptype active generation 0 network-id 2
a=candidate:1081509197 1 tcp 1518214911 192.168.1.52 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=candidate:131530653 2 tcp 1518280446 192.168.1.62 9 typ host tcptype active generation 0 network-id 2
a=candidate:1081509197 2 tcp 1518214910 192.168.1.52 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=candidate:3357345241 1 udp 1686052607 XX.XXX.XX.XX 35116 typ srflx raddr 192.168.1.62 rport 57708 generation 0 network-id 2
a=candidate:3357345241 2 udp 1686052606 XX.XXX.XX.XX 35118 typ srflx raddr 192.168.1.62 rport 57710 generation 0 network-id 2
a=ice-ufrag:CqCq
a=ice-pwd:a1k9+EzlXAZSRKg17sB5bwqs
a=ice-options:trickle
a=fingerprint:sha-256 32:E2:66:71:3F:EE:94:26:EA:B0:22:EB:21:2F:EB:E0:CB:B4:C2:9C:54:30:82:84:D8:19:C6:13:9D:CF:F7:05
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:R7XC1bG60xk5SCT9NcJHV63VrWekNgUfgKnm 7833911e-885c-493f-8dc1-cc86b39201f1
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1867325341 cname:cT1YyFvhOXhlbNhn
a=ssrc:1867325341 msid:R7XC1bG60xk5SCT9NcJHV63VrWekNgUfgKnm 7833911e-885c-493f-8dc1-cc86b39201f1
a=ssrc:1867325341 mslabel:R7XC1bG60xk5SCT9NcJHV63VrWekNgUfgKnm
a=ssrc:1867325341 label:7833911e-885c-493f-8dc1-cc86b39201f1

tsk_utils.js?svn=252:115 __tsip_transport_ws_onmessage
tsk_utils.js?svn=252:115 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=56173;received=192.168.1.62;branch=z9hG4bKNUr3Doml8vlIgTKPuTCkLRhSgTUfAThb
From: "3000"<sip:3000@192.168.1.7>;tag=KIVWD4blxGuIakHafyEu
To: <sip:077XXXXXXXX@NULL>;tag=z9hG4bKNUr3Doml8vlIgTKPuTCkLRhSgTUfAThb
Call-ID: 342413b2-30db-06d8-6b18-fcc16be87a7d
CSeq: 17323 INVITE
Content-Length: 0
WWW-Authenticate: Digest realm="",qop="auth",nonce="1592555593/d6dfb8102688ced5a183007f48171877",opaque="3b4102ed6408101e",stale=FALSE,algorithm=md5
Server: Asterisk PBX 17.3.0


tsk_utils.js?svn=252:115 SEND: ACK sip:077XXXXXXXX@NULL SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKNUr3Doml8vlIgTKPuTCkLRhSgTUfAThb;rport
From: "3000"<sip:3000@192.168.1.7>;tag=KIVWD4blxGuIakHafyEu
To: <sip:077XXXXXXXX@NULL>;tag=z9hG4bKNUr3Doml8vlIgTKPuTCkLRhSgTUfAThb
Call-ID: 342413b2-30db-06d8-6b18-fcc16be87a7d
CSeq: 17323 ACK
Content-Length: 0
Max-Forwards: 70


tsk_utils.js?svn=252:115 State machine: x0000_Any_2_Any_X_i401_407_INVITE
tsk_utils.js?svn=252:115 SEND: INVITE sip:077XXXXXXXX@NULL SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKN7H5iUGROICR1IDfkkm6l7afIGeXOpuE;rport
From: "3000"<sip:3000@192.168.1.7>;tag=KIVWD4blxGuIakHafyEu
To: <sip:077XXXXXXXX@NULL>
Contact: "3000"<sips:3000@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;impi=3000;ha1=b57e29d9ee165b9ee6a9457a734b8cac;+g.oma.sip-im;language="en,fr"
Call-ID: 342413b2-30db-06d8-6b18-fcc16be87a7d
CSeq: 17324 INVITE
Content-Type: application/sdp
Content-Length: 2821
Max-Forwards: 70
Authorization: Digest username="3000",nonce="1592555593/d6dfb8102688ced5a183007f48171877",uri="sip:077XXXXXXXX@NULL",response="043db13bff578fac2e5640ab3408d0ac",algorithm=md5,cnonce="40885d53a1cc454c0ffabeca04f28541",opaque="3b4102ed6408101e",qop=auth,nc=00000001
User-Agent: 199
Organisation: callcent

v=0
o=- 2062517073951919000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS R7XC1bG60xk5SCT9NcJHV63VrWekNgUfgKnm
m=audio 35116 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 XX.XXX.XX.XX
a=rtcp:35118 IN IP4 XX.XXX.XX.XX
a=candidate:1230308205 1 udp 2122260223 192.168.1.62 57708 typ host generation 0 network-id 2
a=candidate:250968509 1 udp 2122194687 192.168.1.52 57709 typ host generation 0 network-id 1 network-cost 10
a=candidate:1230308205 2 udp 2122260222 192.168.1.62 57710 typ host generation 0 network-id 2
a=candidate:250968509 2 udp 2122194686 192.168.1.52 57711 typ host generation 0 network-id 1 network-cost 10
a=candidate:131530653 1 tcp 1518280447 192.168.1.62 9 typ host tcptype active generation 0 network-id 2
a=candidate:1081509197 1 tcp 1518214911 192.168.1.52 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=candidate:131530653 2 tcp 1518280446 192.168.1.62 9 typ host tcptype active generation 0 network-id 2
a=candidate:1081509197 2 tcp 1518214910 192.168.1.52 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=candidate:3357345241 1 udp 1686052607 XX.XXX.XX.XX 35116 typ srflx raddr 192.168.1.62 rport 57708 generation 0 network-id 2
a=candidate:3357345241 2 udp 1686052606 XX.XXX.XX.XX 35118 typ srflx raddr 192.168.1.62 rport 57710 generation 0 network-id 2
a=ice-ufrag:CqCq
a=ice-pwd:a1k9+EzlXAZSRKg17sB5bwqs
a=ice-options:trickle
a=fingerprint:sha-256 32:E2:66:71:3F:EE:94:26:EA:B0:22:EB:21:2F:EB:E0:CB:B4:C2:9C:54:30:82:84:D8:19:C6:13:9D:CF:F7:05
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:R7XC1bG60xk5SCT9NcJHV63VrWekNgUfgKnm 7833911e-885c-493f-8dc1-cc86b39201f1
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1867325341 cname:cT1YyFvhOXhlbNhn
a=ssrc:1867325341 msid:R7XC1bG60xk5SCT9NcJHV63VrWekNgUfgKnm 7833911e-885c-493f-8dc1-cc86b39201f1
a=ssrc:1867325341 mslabel:R7XC1bG60xk5SCT9NcJHV63VrWekNgUfgKnm
a=ssrc:1867325341 label:7833911e-885c-493f-8dc1-cc86b39201f1

tsk_utils.js?svn=252:115 __tsip_transport_ws_onmessage
tsk_utils.js?svn=252:115 recv=SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=56173;received=192.168.1.62;branch=z9hG4bKN7H5iUGROICR1IDfkkm6l7afIGeXOpuE
From: "3000"<sip:3000@192.168.1.7>;tag=KIVWD4blxGuIakHafyEu
To: <sip:077XXXXXXXX@NULL>
Call-ID: 342413b2-30db-06d8-6b18-fcc16be87a7d
CSeq: 17324 INVITE
Content-Length: 0
Server: Asterisk PBX 17.3.0


tsk_utils.js?svn=252:115 State machine: x0000_Any_2_Any_X_i1xx
tsk_utils.js?svn=252:115 ==session event = i_ao_request
tsk_utils.js?svn=252:115 __tsip_transport_ws_onmessage
tsk_utils.js?svn=252:115 recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=56173;received=192.168.1.62;branch=z9hG4bKN7H5iUGROICR1IDfkkm6l7afIGeXOpuE
From: "3000"<sip:3000@192.168.1.7>;tag=KIVWD4blxGuIakHafyEu
To: <sip:077XXXXXXXX@NULL>;tag=3499be6c-f1c3-40ce-bf51-adc344e42b41
Contact: <sips:192.168.1.7:8089;transport=ws>
Call-ID: 342413b2-30db-06d8-6b18-fcc16be87a7d
CSeq: 17324 INVITE
Content-Type: application/sdp
Content-Length: 1016
Server: Asterisk PBX 17.3.0
Allow: OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,PUBLISH,INVITE,ACK,BYE,CANCEL,UPDATE,PRACK,MESSAGE,REFER
Supported: 100rel,timer,replaces,norefersub

v=0
o=- 3759475608 4 IN IP4 192.168.1.7
s=Asterisk
c=IN IP4 192.168.1.7
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 17036 UDP/TLS/RTP/SAVPF 0 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9B:0F:0D:BB:2A:88:01:75:90:3E:AE:BC:0B:C5:9A:B1:97:DD:E8:D3:9A:23:E2:53:36:95:DB:57:2C:AE:E5:52
a=ice-ufrag:2c24fa2d1f57fc45336db03e164aac29
a=ice-pwd:41df29a27d7311067b8372ae7a0c5c0e
a=candidate:Hdef41758 1 UDP 2130706431 fe80::226:55ff:fe29:ad4a 17036 typ host
a=candidate:Hc0a82d03 1 UDP 2130706431 192.168.1.7 17036 typ host
a=candidate:S59d460d 1 UDP 1694498815 XX.XXX.XX.XX 17036 typ srflx raddr 192.168.1.7 rport 17036
a=rtpmap:0 PCMU/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1985794971 cname:91df80e3-58ed-45ab-880a-f2d15ce6f771
a=msid:5c44aca9-ab24-4afd-acaf-e75fc80e3c6b a814770e-8e2c-4e3e-8d39-a507559b6564
a=rtcp-fb:* transport-cc
a=mid:0

tsk_utils.js?svn=252:115 State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE
tsk_utils.js?svn=252:115 setRemoteDescription(answer)
v=0
o=- 3759475608 4 IN IP4 192.168.1.7
s=Asterisk
c=IN IP4 192.168.1.7
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 17036 RTP/SAVPF 0 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9B:0F:0D:BB:2A:88:01:75:90:3E:AE:BC:0B:C5:9A:B1:97:DD:E8:D3:9A:23:E2:53:36:95:DB:57:2C:AE:E5:52
a=ice-ufrag:2c24fa2d1f57fc45336db03e164aac29
a=ice-pwd:41df29a27d7311067b8372ae7a0c5c0e
a=candidate:Hdef41758 1 UDP 2130706431 fe80::226:55ff:fe29:ad4a 17036 typ host
a=candidate:Hc0a82d03 1 UDP 2130706431 192.168.1.7 17036 typ host
a=candidate:S59d460d 1 UDP 1694498815 XX.XXX.XX.XX 17036 typ srflx raddr 192.168.1.7 rport 17036
a=rtpmap:0 PCMU/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1985794971 cname:91df80e3-58ed-45ab-880a-f2d15ce6f771
a=msid:5c44aca9-ab24-4afd-acaf-e75fc80e3c6b a814770e-8e2c-4e3e-8d39-a507559b6564
a=rtcp-fb:* transport-cc
a=mid:0

tsk_utils.js?svn=252:115 SEND: ACK sips:192.168.1.7:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKO17PhVi2OBMr6yLNGyrg;rport
From: "3000"<sip:3000@192.168.1.7>;tag=KIVWD4blxGuIakHafyEu
To: <sip:077XXXXXXXX@NULL>;tag=3499be6c-f1c3-40ce-bf51-adc344e42b41
Contact: "3000"<sips:3000@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 342413b2-30db-06d8-6b18-fcc16be87a7d
CSeq: 17324 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="3000",nonce="1592555593/d6dfb8102688ced5a183007f48171877",uri="sips:192.168.1.7:8089;transport=ws",response="17dbd7cc27fd02269db65a77e66b6ddd",algorithm=md5,cnonce="40885d53a1cc454c0ffabeca04f28541",opaque="3b4102ed6408101e",qop=auth,nc=00000002
User-Agent: 199
Organisation: callcent


tsk_utils.js?svn=252:115 onSignalingstateChange:stable
tsk_utils.js?svn=252:115 ==session event = m_early_media
tsk_utils.js?svn=252:115 ==session event = connected
tsk_utils.js?svn=252:115 __on_add_stream
tsk_utils.js?svn=252:115 onSetRemoteDescriptionSuccess
tsk_utils.js?svn=252:115 ==session event = m_stream_audio_remote_added
tsk_utils.js?svn=252:115 __tsip_transport_ws_onmessage
tsk_utils.js?svn=252:115 State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister
tsk_utils.js?svn=252:115 SEND: REGISTER sip:NULL SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKUsDxcetCEkGd2tDofeN0nqTw3nd6eKTn;rport
From: "3000"<sip:3000@192.168.1.7>;tag=2FQQk5Ree1O80c0yxmCZ
To: "3000"<sip:3000@192.168.1.7>
Contact: "3000"<sips:3000@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 47845c69-5878-9c6f-687d-ce59c579ef3a
CSeq: 15341 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="3000",nonce="1592555526/d700ca2e0d523653b6b07b5ed951a2ee",uri="sip:NULL",response="73509d50241316863838b3b9880c4097",algorithm=md5,cnonce="42dd80965c9492d2d85375d7faca5c8f",opaque="38fbf3f617b8f3ad",qop=auth,nc=00000002
User-Agent: 199
Organisation: callcent


tsk_utils.js?svn=252:115 __tsip_transport_ws_onmessage
tsk_utils.js?svn=252:115 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=56173;received=192.168.1.62;branch=z9hG4bKUsDxcetCEkGd2tDofeN0nqTw3nd6eKTn
From: "3000"<sip:3000@192.168.1.7>;tag=2FQQk5Ree1O80c0yxmCZ
To: "3000"<sip:3000@192.168.1.7>;tag=z9hG4bKUsDxcetCEkGd2tDofeN0nqTw3nd6eKTn
Call-ID: 47845c69-5878-9c6f-687d-ce59c579ef3a
CSeq: 15341 REGISTER
Content-Length: 0
WWW-Authenticate: Digest realm="",qop="auth",nonce="1592555626/25a8da0776649e0934754e84adf31c5c",opaque="06f580a4649197c7",stale=TRUE,algorithm=md5
Server: Asterisk PBX 17.3.0


tsk_utils.js?svn=252:115 State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
tsk_utils.js?svn=252:115 SEND: REGISTER sip:NULL SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKx0b6L8i6D8wNqkVBvawTNVSP35rFGTGX;rport
From: "3000"<sip:3000@192.168.1.7>;tag=2FQQk5Ree1O80c0yxmCZ
To: "3000"<sip:3000@192.168.1.7>
Contact: "3000"<sips:3000@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 47845c69-5878-9c6f-687d-ce59c579ef3a
CSeq: 15342 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="3000",nonce="1592555626/25a8da0776649e0934754e84adf31c5c",uri="sip:NULL",response="be074fc9a81cb9763947b1c9fa261f4c",algorithm=md5,cnonce="195d726d5d64d23975a2ed014acacaa3",opaque="06f580a4649197c7",qop=auth,nc=00000001
User-Agent: 199
Organisation: callcent


tsk_utils.js?svn=252:115 __tsip_transport_ws_onmessage
tsk_utils.js?svn=252:115 recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=56173;received=192.168.1.62;branch=z9hG4bKx0b6L8i6D8wNqkVBvawTNVSP35rFGTGX
From: "3000"<sip:3000@192.168.1.7>;tag=2FQQk5Ree1O80c0yxmCZ
To: "3000"<sip:3000@192.168.1.7>;tag=z9hG4bKx0b6L8i6D8wNqkVBvawTNVSP35rFGTGX
Contact: <sips:3000@192.168.1.62:56173;transport=ws;rtcweb-breaker=yes>;expires=199
Call-ID: 47845c69-5878-9c6f-687d-ce59c579ef3a
CSeq: 15342 REGISTER
Content-Length: 0
Date: 19 Jun 2020 08:33:46 GMT;19
Server: Asterisk PBX 17.3.0


tsk_utils.js?svn=252:115 State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
2tsk_utils.js?svn=252:115 ==session event = sent_request
tsk_utils.js?svn=252:115 __tsip_transport_ws_onmessage
tsk_utils.js?svn=252:115 recv=BYE sips:3000@192.168.1.62:56173;transport=ws;rtcweb-breaker=yes;click2call=no SIP/2.0
Via: SIP/2.0/WSS 192.168.1.7:8089;rport;branch=z9hG4bKPjcfda12fe-dd1d-40f7-84a4-e2aae0b3f6ce;alias
From: <sip:077XXXXXXXX@NULL>;tag=3499be6c-f1c3-40ce-bf51-adc344e42b41
To: "3000"<sip:3000@192.168.1.7>;tag=KIVWD4blxGuIakHafyEu
Call-ID: 342413b2-30db-06d8-6b18-fcc16be87a7d
CSeq: 17682 BYE
Content-Length: 0
Reason: cause=44;cause=44
Max-Forwards: 70
User-Agent: Asterisk PBX 17.3.0


tsk_utils.js?svn=252:115 State machine: x0000_Any_2_Terminated_X_iBYE
tsk_utils.js?svn=252:115 === INVITE Dialog terminated ===
tsk_utils.js?svn=252:115 PeerConnection::stop()
tsk_utils.js?svn=252:115 SEND: SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.168.1.7:8089;rport=8089;branch=z9hG4bKPjcfda12fe-dd1d-40f7-84a4-e2aae0b3f6ce;alias
From: <sip:077XXXXXXXX@NULL>;tag=3499be6c-f1c3-40ce-bf51-adc344e42b41
To: "3000"<sip:3000@192.168.1.7>;tag=KIVWD4blxGuIakHafyEu
Contact: <sips:3000@df7jal23ls0d.invalid;transport=wss>
Call-ID: 342413b2-30db-06d8-6b18-fcc16be87a7d
CSeq: 17682 BYE
Content-Length: 0


tsk_utils.js?svn=252:115 ==session event = terminated

david551 · June 18, 2020, 5:39pm

Please provide sip protocol traces. In particular you need to check the Contact address on the INVITE, and whether BYE is actually being recevied.

ambiorixg12 · June 18, 2020, 6:22pm

Looks BYE request is not received, call keeps open but then get disconnected due to the lack of RTP activity,

david551 · June 18, 2020, 7:12pm

I’m also assuming the BYE isn’t getting through.

ambiorixg12 · June 18, 2020, 9:19pm

I know you do, I just adding some more details

progressaurus · June 19, 2020, 9:03am

Thanks David, I’ve appended the log to the original post. The BYE is’nt being recieved when I hangup the call from the external mobile device. Only when the timeout is triggered does the BYE message generate.

progressaurus · June 19, 2020, 11:05am

Im seeing the below in the sip contact address on the INVITE which doesnt look correct, and looking into it just now:

df7jal23ls0d.invalid

david551 · June 19, 2020, 11:16am

I noticed that, but I’m not sure that it is actually used for WSS, which is why I didn’t comment on it, as I know very little about WSS. For UDP, it would definitely explain the problem.

jcolp · June 19, 2020, 12:36pm

With WSS that’s perfectly fine. The client establishes a connection to Asterisk, and that connection has to be reused for everything. Since the client likely doesn’t know it’s IP, and it doesn’t matter, it places an invalid dummy string on the Contact.

david551 · June 19, 2020, 12:48pm

I’d guess that either the WSS connection has been taken down, or a router has deleted a temporary firewall or NAT rule, with the result that the connection is effectively blocked.

progressaurus · June 19, 2020, 1:17pm

There are two asterisk boxes behind the same NAT the one I am using to test the web phone using pjsip and a production one using chan SIP. Both use carriers which use ip authentication. The router is configured to route the trafic from each provider to the correct box. Could this potentially be causing an issue?

In the meantime I’ve asked the supplier to check if there are any issues on their side and carrying out some logging on the router to hopefully shed some light.

progressaurus · July 13, 2020, 9:06am

Thank you for your input on this gents. I have since tried two other suppliers both of whom worked with no issues. I took this back to the original supplier and explained that my configuration was working fine with two other suppliers. They resonded with this…

“We do a few different types of SIP with the main two being carrier SIP and pbx SIP. The Turtle proxies are the direct carrier SIP and don’t really have any intelligence around traffic/packet management. The proxy below is on the pbx SIP and has a degree of traffic management to allow users to connect with less config.”

So although my config seems to be ok, there was no actual solution to the issue with the original trunk.

Thanks

system · August 12, 2020, 9:06am

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