SIP messages when making a call to local extension

SIP Debugging when making a call to local extension
The extension only plays a "hello world" recording


INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK2696832
Max-Forwards: 69
To: <sip:[email protected]>
From: "UA WebRTC" <sip:[email protected]>;tag=9q1k7ijiab
Call-ID: 6gejrmupvj5asi2fdhv8
CSeq: 9260 INVITE
Contact: <sip:[email protected];transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 0.7.4
Content-Length: 936

v=0
o=mozilla...THIS_IS_SDPARTA-40.0 4294967295 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 2F:5D:E3:B3:9B:9C:8E:00:AF:81:CC:E8:39:CA:05:CF:40:D1:66:11:91:3A:F1:A2:6D:78:D5:51:5F:22:96:AC
a=group:BUNDLE sdparta_0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 35823 RTP/SAVPF 109 9 0 8
c=IN IP4 192.168.0.4
a=candidate:0 1 UDP 2122252543 192.168.0.4 35823 typ host
a=candidate:0 2 UDP 2122252542 192.168.0.4 35135 typ host
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=ice-pwd:516e745ffd8bf121177f0b7680506018
a=ice-ufrag:37f2205c
a=mid:sdparta_0
a=msid:{aa42a325-b228-4ff3-b52f-de1f1288836e} {7b2300a8-f4ea-4b74-bbfa-ac5c4835b3c8}
a=rtcp:35135 IN IP4 192.168.0.4
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:1479786588 cname:{0dbce354-bdf6-4ee7-acb0-e4f12c501e04}

 +70ms
JsSIP:Transport
received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK2696832;received=192.168.0.4
From: "UA WebRTC" <sip:[email protected]>;tag=9q1k7ijiab
To: <sip:[email protected]>;tag=as34ac1926
Call-ID: 6gejrmupvj5asi2fdhv8
CSeq: 9260 INVITE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d67a76d"
Content-Length: 0

 +43ms
JsSIP:Transport
sending WebSocket message:

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK2696832
To: <sip:[email protected]>;tag=as34ac1926
From: "UA WebRTC" <sip:[email protected]>;tag=9q1k7ijiab
Call-ID: 6gejrmupvj5asi2fdhv8
CSeq: 9260 ACK
Content-Length: 0

 +40ms
JsSIP:Transport
sending WebSocket message:

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK3632996
Max-Forwards: 69
To: <sip:[email protected]>
From: "UA WebRTC" <sip:[email protected]>;tag=9q1k7ijiab
Call-ID: 6gejrmupvj5asi2fdhv8
CSeq: 9261 INVITE
Authorization: Digest algorithm=MD5, username="1001", realm="asterisk", nonce="3d67a76d", uri="sip:[email protected]", response="abff34d15fffc9b5baaadeab85d770c9"
Contact: <sip:[email protected];transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 0.7.4
Content-Length: 936

v=0
o=mozilla...THIS_IS_SDPARTA-40.0 4294967295 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 2F:5D:E3:B3:9B:9C:8E:00:AF:81:CC:E8:39:CA:05:CF:40:D1:66:11:91:3A:F1:A2:6D:78:D5:51:5F:22:96:AC
a=group:BUNDLE sdparta_0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 35823 RTP/SAVPF 109 9 0 8
c=IN IP4 192.168.0.4
a=candidate:0 1 UDP 2122252543 192.168.0.4 35823 typ host
a=candidate:0 2 UDP 2122252542 192.168.0.4 35135 typ host
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=ice-pwd:516e745ffd8bf121177f0b7680506018
a=ice-ufrag:37f2205c
a=mid:sdparta_0
a=msid:{aa42a325-b228-4ff3-b52f-de1f1288836e} {7b2300a8-f4ea-4b74-bbfa-ac5c4835b3c8}
a=rtcp:35135 IN IP4 192.168.0.4
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:1479786588 cname:{0dbce354-bdf6-4ee7-acb0-e4f12c501e04}

 +18ms
JsSIP:InviteClientTransaction Timer D expired for transaction z9hG4bK2696832 +44ms
JsSIP:Transport
received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK3632996;received=192.168.0.4
From: "UA WebRTC" <sip:[email protected]>;tag=9q1k7ijiab
To: <sip:[email protected]>
Call-ID: 6gejrmupvj5asi2fdhv8
CSeq: 9261 INVITE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:[email protected]:5060;transport=WS>
Content-Length: 0

 +116ms
JsSIP:RTCSession receiveInviteResponse() +31ms
JsSIP:Transport
received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK3632996;received=192.168.0.4
From: "UA WebRTC" <sip:[email protected]>;tag=9q1k7ijiab
To: <sip:[email protected]>;tag=as2a14b282
Call-ID: 6gejrmupvj5asi2fdhv8
CSeq: 9261 INVITE
Server: Asterisk PBX 12.8.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:[email protected]:5060;transport=WS>
Content-Type: application/sdp
Require: timer
Content-Length: 774

v=0
o=root 1590611571 1590611571 IN IP4 192.168.0.4
s=Asterisk PBX 12.8.2
c=IN IP4 192.168.0.4
t=0 0
m=audio 19864 RTP/SAVPF 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=ice-ufrag:00b9259c164316017df96bd7099a0b59
a=ice-pwd:3eac7a773d965d021529752477b9278e
a=candidate:Hc0a80004 1 UDP 2130706431 192.168.0.4 19864 typ host
a=candidate:Sbdd9418e 1 UDP 1694498815 189.217.65.142 35710 typ srflx raddr 192.168.0.4 rport 19864
a=candidate:Hc0a80004 2 UDP 2130706430 192.168.0.4 19865 typ host
a=candidate:Sbdd9418e 2 UDP 1694498814 189.217.65.142 46555 typ srflx raddr 192.168.0.4 rport 19865
a=connection:new
a=setup:active
a=fingerprint:SHA-256 E2:B9:77:44:04:96:8D:D9:EA:67:B9:E0:6E:93:49:08:5B:33:53:7D:DD:51:5A:1B:0F:7D:45:CA:BD:F9:CE:BA
a=sendrecv

 +8ms
JsSIP:RTCSession receiveInviteResponse() +21ms
JsSIP:Dialog new UAC dialog created with status CONFIRMED +9ms
JsSIP:RTCSession session accepted +29ms
JsSIP:RTCSession sendRequest() +10ms
JsSIP:RTCSession:Request new | ACK +11ms
JsSIP:Transport
sending WebSocket message:

ACK sip:[email protected]:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK5815096
Max-Forwards: 69
To: <sip:[email protected]>;tag=as2a14b282
From: "UA WebRTC" <sip:[email protected]>;tag=9q1k7ijiab
Call-ID: 6gejrmupvj5asi2fdhv8
CSeq: 9261 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
Supported: outbound
User-Agent: JsSIP 0.7.4
Content-Length: 0

 +18ms
JsSIP:RTCSession session confirmed +11ms
 LocalMediaStream { id="{aa42a325-b228-4ff3-b52f-de1f1288836e}",  currentTime=0.300375,  stop=stop(),  more...}//printing of local stream
miScript.js (line 69)
CALL CONFIRMED
miScript.js (line 70)
element is null
jssip-0.7.4.js (line 22924)
 MediaStream { id="{38caf558-e986-4c52-b9cd-4abe5384051e}",  currentTime=0.0309375,  getAudioTracks=getAudioTracks(),  more...}
//printing of remote stream
miScript.js (line 77)
REMOTE STREAM RECEIVED
miScript.js (line 78)
TypeError: element is null
jssip-0.7.4.js (line 22924, col 17)
JsSIP:Transport
received WebSocket text message:

BYE sip:[email protected];transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.0.4:5060;branch=z9hG4bK4a805194
Max-Forwards: 70
From: <sip:[email protected]>;tag=as2a14b282
To: "UA WebRTC" <sip:[email protected]>;tag=9q1k7ijiab
Call-ID: 6gejrmupvj5asi2fdhv8
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.8.2
Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:192.168.0.4", nonce="3d67a76d", response="4ff114fe1b3b95781c646407da538133"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

 +2s
JsSIP:RTCSession receiveRequest() +22ms
JsSIP:Transport
sending WebSocket message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.0.4:5060;branch=z9hG4bK4a805194
To: "UA WebRTC" <sip:[email protected]>;tag=9q1k7ijiab
From: <sip:[email protected]>;tag=as2a14b282
Call-ID: 6gejrmupvj5asi2fdhv8
CSeq: 102 BYE
Supported: outbound
Content-Length: 0

 +12ms
JsSIP:RTCSession session ended +12ms
JsSIP:RTCSession close() +8ms
JsSIP:RTCSession close() | closing local MediaStream +12ms
JsSIP:Dialog dialog 6gejrmupvj5asi2fdhv89q1k7ijiabas2a14b282 deleted +11ms
JsSIP:NonInviteServerTransaction Timer J expired for transaction z9hG4bK4a805194 +29ms
TypeError: e.data is undefined
miScript.js (line 65, col 12)