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- SIP Debugging when making a call to local extension
- The extension only plays a "hello world" recording
- INVITE sip:200@192.168.0.4 SIP/2.0
- Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK2696832
- Max-Forwards: 69
- To: <sip:200@192.168.0.4>
- From: "UA WebRTC" <sip:1001@192.168.0.4>;tag=9q1k7ijiab
- Call-ID: 6gejrmupvj5asi2fdhv8
- CSeq: 9260 INVITE
- Contact: <sip:grd0bbtt@192.0.2.177;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:1001@192.168.0.4>;tag=9q1k7ijiab
- To: <sip:200@192.168.0.4>;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:200@192.168.0.4 SIP/2.0
- Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK2696832
- To: <sip:200@192.168.0.4>;tag=as34ac1926
- From: "UA WebRTC" <sip:1001@192.168.0.4>;tag=9q1k7ijiab
- Call-ID: 6gejrmupvj5asi2fdhv8
- CSeq: 9260 ACK
- Content-Length: 0
- +40ms
- JsSIP:Transport
- sending WebSocket message:
- INVITE sip:200@192.168.0.4 SIP/2.0
- Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK3632996
- Max-Forwards: 69
- To: <sip:200@192.168.0.4>
- From: "UA WebRTC" <sip:1001@192.168.0.4>;tag=9q1k7ijiab
- Call-ID: 6gejrmupvj5asi2fdhv8
- CSeq: 9261 INVITE
- Authorization: Digest algorithm=MD5, username="1001", realm="asterisk", nonce="3d67a76d", uri="sip:200@192.168.0.4", response="abff34d15fffc9b5baaadeab85d770c9"
- Contact: <sip:grd0bbtt@192.0.2.177;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:1001@192.168.0.4>;tag=9q1k7ijiab
- To: <sip:200@192.168.0.4>
- 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:200@192.168.0.4:5060;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:1001@192.168.0.4>;tag=9q1k7ijiab
- To: <sip:200@192.168.0.4>;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:200@192.168.0.4:5060;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:200@192.168.0.4:5060;transport=ws SIP/2.0
- Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK5815096
- Max-Forwards: 69
- To: <sip:200@192.168.0.4>;tag=as2a14b282
- From: "UA WebRTC" <sip:1001@192.168.0.4>;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:grd0bbtt@192.0.2.177;transport=ws;ob SIP/2.0
- Via: SIP/2.0/WS 192.168.0.4:5060;branch=z9hG4bK4a805194
- Max-Forwards: 70
- From: <sip:200@192.168.0.4>;tag=as2a14b282
- To: "UA WebRTC" <sip:1001@192.168.0.4>;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:1001@192.168.0.4>;tag=9q1k7ijiab
- From: <sip:200@192.168.0.4>;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)
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