Asterisk + JsSIP + Amazon EC2

system · February 17, 2015, 3:28pm

Good day, I try to implement a bunch of Asterisk + JsSip on Amazon EC2 to make a call from browser to browser. As you know Amazon is behind NAT and its external (Elastic IP) varies with the internal. To solve this problem,I’m must use STUN or rather ICE. So, I am conigure Asterisk 11.13.1 + srtp. created to test several accounts, the call for SIP using a standard 3CX Phone passes (And no, without directmedia, canreinvite and other functionality which is trying to force customers to call each other on a straight line) call is precisely through the server console in debug RTP perfectly see running around traffic, as well as the usual tcpdump. But when I’m try to call using WebRTC media stream in the form of RTP I do not get, respectively, and the voice, too … Although if I’m calling from a softphone on the web that RTP between hosts I see. Think my problem is with ICE but where to look … some help please, enclose their configs

 rtp.conf



[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=stun.l.google.com:19302

sip.conf

[general]
udpbindaddr=0.0.0.0:5060
realm=myhostname
transport=udp,ws,wss
externip=myextip
context=realtime
rtcachefriends=yes
rtsavesysname=yes
language=en
localnet=172.0.0.0/20
nat=force_rport,comedia
icesupport=yes
accept_outofcall_message=yes
outofcall_message_context=messages
;auth_message_requests = no

[101]
type=friend
username=101
host=dynamic
secret=101
icesupport=no
nat=comedia
qualify=yes
context=testcontext
disallow=all
allow=ulaw
allow=alaw
directmedia=nonat
canreinvite=no

[102]
type=friend
username=102
host=dynamic
secret=102
icesupport=no
nat=comedia
qualify=yes
context=test
disallow=all
allow=ulaw
allow=alaw
directmedia=nonat
canreinvite=no

[777]
type=friend
username=777
host=dynamic
secret=777
encryption=yes
avpf=yes
icesupport=yes
nat=force_rport,comedia
context=testcontext
directmedia=nonat
canreinvite=no
transport=ws,udp
force_avp=yes
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
srtpcapable=yes
disallow=all
allow=ulaw
allow=alaw

[888]
type=friend
username=888
host=dynamic
secret=888
encryption=yes
avpf=yes
icesupport=yes
nat=force_rport,comedia
context=testcontext
directmedia=nonat
canreinvite=no
transport=ws,udp
force_avp=yes
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
srtpcapable=yes
disallow=all
allow=ulaw
allow=alaw

users 101 and 102 are designed to test for SIP, 777 and 888 users who are trying to use WebRTC.
There is web debug:

888:
voip.js?v10:25 init
jssip-0.6.18.js:21376 JsSIP version 0.6.18 +0ms
jssip-0.6.18.js:21376 rtcninja version 0.5.0 +6ms
jssip-0.6.18.js:21376 rtcninja detected browser: Chrome 39.0 [mobile:false, tablet:false, android:false, ios:false] +2ms
jssip-0.6.18.js:21376 JsSIP:UA configuration parameters after validation: +66ms
jssip-0.6.18.js:21376 JsSIP:UA - via_host: "192.0.2.68" +1ms
jssip-0.6.18.js:21376 JsSIP:UA - password: NOT SHOWN +0ms
jssip-0.6.18.js:21376 JsSIP:UA - register_expires: 600 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - register: true +0ms
jssip-0.6.18.js:21376 JsSIP:UA - registrar_server: sip:54.200.180.31 +1ms
jssip-0.6.18.js:21376 JsSIP:UA - ws_server_max_reconnection: 3 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - ws_server_reconnection_timeout: 4 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - connection_recovery_min_interval: 2 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - connection_recovery_max_interval: 30 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - use_preloaded_route: false +0ms
jssip-0.6.18.js:21376 JsSIP:UA - no_answer_timeout: 60000 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - session_timers: false +0ms
jssip-0.6.18.js:21376 JsSIP:UA - hack_via_tcp: false +1ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - hack_via_ws: false +0ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - hack_ip_in_contact: true +0ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - node_websocket_options: {} +0ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - uri: sip:888@54.200.180.31 +1ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - ws_servers: [{"ws_uri":"ws://54.200.180.31:8088/ws","sip_uri":"<sip:54.200.180.31:8088;transport=ws;lr>","weight":0,"status":0,"scheme":"WS"}] +0ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - display_name: "888" +0ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - instance_id: "0afee81f-ce1f-41a2-b377-d3e0a4c2b1d3" +0ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - jssip_id: "2inrp" +0ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - hostport_params: "54.200.180.31" +1ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA - authorization_user: "888" +0ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 rtcninja WebRTC supported +1ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:UA start() +1ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:Transport connecting to WebSocket ws://54.200.180.31:8088/ws +1ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:Transport WebSocket ws://54.200.180.31:8088/ws connected +451ms
voip.js:268 message Connection success
voip.js:25 ready
voip.js:764 UA
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

REGISTER sip:54.200.180.31 SIP/2.0
Via: SIP/2.0/WS 192.0.2.68;branch=z9hG4bK5160676
Max-Forwards: 69
To: <sip:888@54.200.180.31>
From: "888" <sip:888@54.200.180.31>;tag=h5o5autqle
Call-ID: uv31655tk91b6f24havq4v
CSeq: 1 REGISTER
Contact: <sip:kk2q0bsv@192.0.2.68;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0afee81f-ce1f-41a2-b377-d3e0a4c2b1d3>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: path,gruu,outbound
User-Agent: JsSIP 0.6.18
Content-Length: 0


 +10ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 192.0.2.68;branch=z9hG4bK5160676;received=178.219.174.6;rport=42908
From: "888" <sip:888@54.200.180.31>;tag=h5o5autqle
To: <sip:888@54.200.180.31>;tag=as612041cf
Call-ID: uv31655tk91b6f24havq4v
CSeq: 1 REGISTER
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="54.200.180.31", nonce="3835a6a1"
Content-Length: 0


 +228ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

REGISTER sip:54.200.180.31 SIP/2.0
Via: SIP/2.0/WS 192.0.2.68;branch=z9hG4bK2287834
Max-Forwards: 69
To: <sip:888@54.200.180.31>
From: "888" <sip:888@54.200.180.31>;tag=h5o5autqle
Call-ID: uv31655tk91b6f24havq4v
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="888", realm="54.200.180.31", nonce="3835a6a1", uri="sip:54.200.180.31", response="eaba67db026a42f0d1b69f132710c877"
Contact: <sip:kk2q0bsv@192.0.2.68;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0afee81f-ce1f-41a2-b377-d3e0a4c2b1d3>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: path,gruu,outbound
User-Agent: JsSIP 0.6.18
Content-Length: 0


 +22ms
:59880/voip/js/jssip/jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.68;branch=z9hG4bK2287834;received=178.219.174.6;rport=42908
From: "888" <sip:888@54.200.180.31>;tag=h5o5autqle
To: <sip:888@54.200.180.31>;tag=as612041cf
Call-ID: uv31655tk91b6f24havq4v
CSeq: 2 REGISTER
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:kk2q0bsv@192.0.2.68;transport=ws>;expires=600
Date: Tue, 17 Feb 2015 14:26:23 GMT
Content-Length: 0


 +227ms
voip.js:268 message Registered
voip.js:25 status.active
voip.js?v10:25 status.inactive
jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

INVITE sip:kk2q0bsv@192.0.2.68;transport=ws SIP/2.0
Via: SIP/2.0/WS 54.200.180.31:0;branch=z9hG4bK22c34220;rport
Max-Forwards: 70
From: "777" <sip:777@54.200.180.31:0>;tag=as1d66f2c7
To: <sip:kk2q0bsv@192.0.2.68;transport=ws>
Contact: <sip:777@54.200.180.31:0;transport=WS>
Call-ID: 7288e1c61bb4bb33643429d376828d68@54.200.180.31:0
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1
Date: Tue, 17 Feb 2015 14:26:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 809

v=0
o=root 760910156 760910156 IN IP4 54.200.180.31
s=Asterisk PBX 11.13.1
c=IN IP4 54.200.180.31
t=0 0
m=audio 18942 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:6eef9bfc044636cf4a2ed49137f09e82
a=ice-pwd:0c139f73247bfcc61fd76bd924c2c2bc
a=candidate:Hac1f0f11 1 UDP 2130706431 172.31.15.17 18942 typ host
a=candidate:S36c8b41f 1 UDP 1694498815 54.200.180.31 18942 typ srflx
a=candidate:Hac1f0f11 2 UDP 2130706430 172.31.15.17 18943 typ host
a=candidate:S36c8b41f 2 UDP 1694498814 54.200.180.31 18943 typ srflx
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 34:1F:33:1C:B5:14:3D:D9:F4:25:7E:29:60:96:8A:61:53:F0:B9:B4:47:1E:A5:E4:5A:82:DC:A5:1E:FF:1E:94
a=sendrecv

 +9s
jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS 54.200.180.31:0;branch=z9hG4bK22c34220;rport
To: <sip:kk2q0bsv@192.0.2.68;transport=ws>
From: "777" <sip:777@54.200.180.31:0>;tag=as1d66f2c7
Call-ID: 7288e1c61bb4bb33643429d376828d68@54.200.180.31:0
CSeq: 102 INVITE
Supported: ice,outbound
Content-Length: 0


 +17ms
jssip-0.6.18.js:21376 JsSIP:RTCSession new +1ms
jssip-0.6.18.js:21376 JsSIP:RTCSession init_incoming() +1ms
jssip-0.6.18.js:21376 JsSIP:Dialog new UAS dialog created with status EARLY +1ms
jssip-0.6.18.js:21376 JsSIP:RTCSession newRTCSession +1ms
voip.js?v10:262 debug newRTCSession Object {originator: "remote", session: RTCSession, request: IncomingRequest}
voip.js?v10:262 debug add session RTCSession {ua: UA, status: 4, dialog: null, earlyDialogs: Object, connection: null…}
voip.js?v10:268 message Incoming call from  777
voip.js?v10:25 incomingCall.wait
jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

SIP/2.0 180 Ringing
Via: SIP/2.0/WS 54.200.180.31:0;branch=z9hG4bK22c34220;rport
To: <sip:kk2q0bsv@192.0.2.68;transport=ws>;tag=l4r7likalc
From: "777" <sip:777@54.200.180.31:0>;tag=as1d66f2c7
Call-ID: 7288e1c61bb4bb33643429d376828d68@54.200.180.31:0
CSeq: 102 INVITE
Contact: <sip:kk2q0bsv@192.0.2.68;transport=ws>
Supported: ice,outbound
Content-Length: 0


 +13ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session progress +0ms
voip.js?v10:25 answer.before
voip.js?v10:262 debug answer RTCSession {ua: UA, status: 4, dialog: null, earlyDialogs: Object, connection: null…}
jssip-0.6.18.js:21376 JsSIP:RTCSession answer() +2s
jssip-0.6.18.js:21376 JsSIP:Dialog dialog 7288e1c61bb4bb33643429d376828d68@54.200.180.31:0l4r7likalcas1d66f2c7  changed to CONFIRMED state +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection new | pcConfig: +5ms Object {iceServers: Array[1]}
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setConfigurationAndOptions | processed pcConfig: +2ms Object {iceServers: Array[1]}
jssip-0.6.18.js:21376 rtcninja:Adapter getUserMedia() | constraints: +7ms Object {audio: true, video: false}
voip.js?v10:25 answer
voip.js?v10:25 incomingCall.connect
jssip-0.6.18.js:21376 rtcninja:Adapter getUserMedia() | success +1s
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection addStream() | stream: +1ms MediaStream {onremovetrack: null, onaddtrack: null, onended: null, ended: false, id: "EyY0Tc0pJ65tdsLIXubTvwT4bLn4j5G4PIXt"…}
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setRemoteDescription() +3ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onnegotiationneeded() +4ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: have-remote-offer +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onaddstream() | stream: +1ms MediaStream {onremovetrack: null, onaddtrack: null, onended: null, ended: false, id: "default"…}
voip.js?v10:262 debug addstream Object {stream: MediaStream}
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setRemoteDescription() | success +3ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session connecting +1ms
voip.js?v10:268 message Connecting with user...
jssip-0.6.18.js:21376 JsSIP:RTCSession createLocalDescription() +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection createAnswer() +2ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection createAnswer() | success +8ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setLocalDescription() +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: stable +4ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection oniceconnectionstatechange() | iceConnectionState: checking +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setLocalDescription() | success +2ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:3859917557 1 udp 2122260223 192.168.1.96 46996 typ host generation 0 +5ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:3859917557 2 udp 2122260222 192.168.1.96 44342 typ host generation 0 +2ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:268555041 2 udp 1686052606 178.219.174.6 44342 typ srflx raddr 192.168.1.96 rport 44342 generation 0 +48ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:268555041 1 udp 1686052607 178.219.174.6 46996 typ srflx raddr 192.168.1.96 rport 46996 generation 0 +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:2828162565 1 tcp 1518280447 192.168.1.96 0 typ host tcptype active generation 0 +43ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:2828162565 2 tcp 1518280446 192.168.1.96 0 typ host tcptype active generation 0 +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | end of candidates +49ms
jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 54.200.180.31:0;branch=z9hG4bK22c34220;rport
To: <sip:kk2q0bsv@192.0.2.68;transport=ws>;tag=l4r7likalc
From: "777" <sip:777@54.200.180.31:0>;tag=as1d66f2c7
Call-ID: 7288e1c61bb4bb33643429d376828d68@54.200.180.31:0
CSeq: 102 INVITE
Contact: <sip:kk2q0bsv@192.0.2.68;transport=ws>
Supported: ice,outbound
Content-Type: application/sdp
Content-Length: 1361

v=0
o=- 2838487268634674867 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS EyY0Tc0pJ65tdsLIXubTvwT4bLn4j5G4PIXt
m=audio 46996 RTP/SAVPF 0 8 101
c=IN IP4 178.219.174.6
a=rtcp:44342 IN IP4 178.219.174.6
a=candidate:3859917557 1 udp 2122260223 192.168.1.96 46996 typ host generation 0
a=candidate:3859917557 2 udp 2122260222 192.168.1.96 44342 typ host generation 0
a=candidate:268555041 2 udp 1686052606 178.219.174.6 44342 typ srflx raddr 192.168.1.96 rport 44342 generation 0
a=candidate:268555041 1 udp 1686052607 178.219.174.6 46996 typ srflx raddr 192.168.1.96 rport 46996 generation 0
a=candidate:2828162565 1 tcp 1518280447 192.168.1.96 0 typ host tcptype active generation 0
a=candidate:2828162565 2 tcp 1518280446 192.168.1.96 0 typ host tcptype active generation 0
a=ice-ufrag:NMOpTzsqDzC8nTe9
a=ice-pwd:sxxTmu6xPcU/5CCuGadJyQx2
a=fingerprint:sha-256 9A:C1:2D:20:4C:16:6A:C7:A4:17:06:E8:02:14:48:25:EA:A3:37:5D:F4:34:E9:76:2C:83:75:C2:C5:63:5A:0B
a=setup:active
a=mid:audio
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:4004630173 cname:K7wrCyJqVTInewVD
a=ssrc:4004630173 msid:EyY0Tc0pJ65tdsLIXubTvwT4bLn4j5G4PIXt f29fcfc2-3ff5-4535-995f-6b108c7c7fba
a=ssrc:4004630173 mslabel:EyY0Tc0pJ65tdsLIXubTvwT4bLn4j5G4PIXt
a=ssrc:4004630173 label:f29fcfc2-3ff5-4535-995f-6b108c7c7fba

 +3ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session accepted +2ms
voip.js?v10:262 debug call accepted Object {originator: "local", response: null}
jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

ACK sip:kk2q0bsv@192.0.2.68;transport=ws SIP/2.0
Via: SIP/2.0/WS 54.200.180.31:0;branch=z9hG4bK21d51b89;rport
Max-Forwards: 70
From: "777" <sip:777@54.200.180.31:0>;tag=as1d66f2c7
To: <sip:kk2q0bsv@192.0.2.68;transport=ws>;tag=l4r7likalc
Contact: <sip:777@54.200.180.31:0;transport=WS>
Call-ID: 7288e1c61bb4bb33643429d376828d68@54.200.180.31:0
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1
Content-Length: 0


 +355ms
jssip-0.6.18.js:21376 JsSIP:RTCSession receiveRequest() +8ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session confirmed +1ms
voip.js?v10:262 debug call confirmed Object {originator: "remote", ack: IncomingRequest}
voip.js?v10:268 message Speaking...
voip.js?v10:25 incomingCall.speak
voip.js?v10:712 call confirmed
voip.js?v10:268 777: 00:00:01
voip.js?v10:268 777: 00:00:02
voip.js?v10:268 777: 00:00:03
voip.js?v10:268 777: 00:00:04
voip.js?v10:268 777: 00:00:05
voip.js?v10:268 777: 00:00:06
voip.js?v10:268 777: 00:00:07
voip.js?v10:268 777: 00:00:08
voip.js?v10:268 777: 00:00:09
voip.js?v10:268 777: 00:00:10
voip.js?v10:25 hangUp.before
voip.js?v10:262 debug hang up RTCSession {ua: UA, status: 9, dialog: Dialog, earlyDialogs: Object, connection: RTCPeerConnection…}
jssip-0.6.18.js:21376 JsSIP:RTCSession terminate() +10s
jssip-0.6.18.js:21376 JsSIP:RTCSession terminating session +0ms
jssip-0.6.18.js:21376 JsSIP:RTCSession sendRequest() +1ms
jssip-0.6.18.js:21376 JsSIP:RTCSession:Request new | BYE +0ms
jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

BYE sip:777@54.200.180.31:0;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.0.2.68;branch=z9hG4bK8905896
Max-Forwards: 69
To: <sip:777@54.200.180.31:0>;tag=as1d66f2c7
From: "888" <sip:kk2q0bsv@192.0.2.68;transport=ws>;tag=l4r7likalc
Call-ID: 7288e1c61bb4bb33643429d376828d68@54.200.180.31:0
CSeq: 9790 BYE
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: outbound
User-Agent: JsSIP 0.6.18
Content-Length: 0


 +4ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session ended +0ms
jssip-0.6.18.js:21376 JsSIP:RTCSession close() +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection close() +1ms
jssip-0.6.18.js:21376 JsSIP:RTCSession close() | closing local MediaStream +2ms
jssip-0.6.18.js:21376 rtcninja:Adapter closeMediaStream() | calling stop() on all the MediaStreamTrack +1ms
jssip-0.6.18.js:21376 JsSIP:Dialog dialog 7288e1c61bb4bb33643429d376828d68@54.200.180.31:0l4r7likalcas1d66f2c7 deleted +1ms
voip.js?v10:262 debug call ended Object {originator: "local", message: null, cause: "Rejected"}
voip.js?v10:268 message Call ended
voip.js?v10:25 incomingCall.end
voip.js?v10:25 hangUp
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection oniceconnectionstatechange() | iceConnectionState: closed +3ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: closed +0ms
jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.68;branch=z9hG4bK8905896;received=178.219.174.6;rport=42908
From: "888" <sip:kk2q0bsv@192.0.2.68;transport=ws>;tag=l4r7likalc
To: <sip:777@54.200.180.31:0>;tag=as1d66f2c7
Call-ID: 7288e1c61bb4bb33643429d376828d68@54.200.180.31:0
CSeq: 9790 BYE
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


 +218ms
jssip-0.6.18.js:21376 JsSIP:RTCSession:Request onSuccessResponse +9ms
jssip-0.6.18.js:21376 JsSIP:InviteServerTransaction Timer L expired for transaction z9hG4bK22c34220 +21s


777:

voip.js?v10:25 init
jssip-0.6.18.js:21376 JsSIP version 0.6.18 +0ms
jssip-0.6.18.js:21376 rtcninja version 0.5.0 +7ms
jssip-0.6.18.js:21376 rtcninja detected browser: Chrome 39.0 [mobile:false, tablet:false, android:false, ios:false] +1ms
jssip-0.6.18.js:21376 JsSIP:UA configuration parameters after validation: +56ms
jssip-0.6.18.js:21376 JsSIP:UA - via_host: "192.0.2.190" +1ms
jssip-0.6.18.js:21376 JsSIP:UA - password: NOT SHOWN +0ms
jssip-0.6.18.js:21376 JsSIP:UA - register_expires: 600 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - register: false +0ms
jssip-0.6.18.js:21376 JsSIP:UA - registrar_server: sip:54.200.180.31 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - ws_server_max_reconnection: 3 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - ws_server_reconnection_timeout: 4 +1ms
jssip-0.6.18.js:21376 JsSIP:UA - connection_recovery_min_interval: 2 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - connection_recovery_max_interval: 30 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - use_preloaded_route: false +0ms
jssip-0.6.18.js:21376 JsSIP:UA - no_answer_timeout: 60000 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - session_timers: false +0ms
jssip-0.6.18.js:21376 JsSIP:UA - hack_via_tcp: false +0ms
jssip-0.6.18.js:21376 JsSIP:UA - hack_via_ws: false +0ms
jssip-0.6.18.js:21376 JsSIP:UA - hack_ip_in_contact: true +0ms
jssip-0.6.18.js:21376 JsSIP:UA - node_websocket_options: {} +1ms
jssip-0.6.18.js:21376 JsSIP:UA - uri: sip:777@54.200.180.31 +0ms
jssip-0.6.18.js:21376 JsSIP:UA - ws_servers: [{"ws_uri":"ws://54.200.180.31:8088/ws","sip_uri":"<sip:54.200.180.31:8088;transport=ws;lr>","weight":0,"status":0,"scheme":"WS"}] +0ms
jssip-0.6.18.js:21376 JsSIP:UA - display_name: "777" +0ms
jssip-0.6.18.js:21376 JsSIP:UA - instance_id: "bc14a359-7a87-4125-b1ec-c5da4d22b1a8" +0ms
jssip-0.6.18.js:21376 JsSIP:UA - jssip_id: "vm16c" +0ms
jssip-0.6.18.js:21376 JsSIP:UA - hostport_params: "54.200.180.31" +0ms
jssip-0.6.18.js:21376 JsSIP:UA - authorization_user: "777" +1ms
jssip-0.6.18.js:21376 rtcninja WebRTC supported +1ms
jssip-0.6.18.js:21376 JsSIP:UA start() +1ms
jssip-0.6.18.js:21376 JsSIP:Transport connecting to WebSocket ws://54.200.180.31:8088/ws +0ms
jssip-0.6.18.js:21376 JsSIP:Transport WebSocket ws://54.200.180.31:8088/ws connected +449ms
voip.js?v10:268 message Connection success
voip.js?v10:25 ready
voip.js?v10:25 callUp.before
voip.js?v10:262 callUp_before 888
voip.js?v10:268 message Give access to your microphone
jssip-0.6.18.js:21376 JsSIP:UA call() +4ms
jssip-0.6.18.js:21376 JsSIP:RTCSession new +1ms
jssip-0.6.18.js:21376 JsSIP:RTCSession connect() +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection new | pcConfig: +9ms Object
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setConfigurationAndOptions | processed pcConfig: +1ms Object
jssip-0.6.18.js:21376 JsSIP:RTCSession newRTCSession +1ms
voip.js?v10:262 debug newRTCSession Object
voip.js?v10:262 debug add session RTCSession
jssip-0.6.18.js:21376 rtcninja:Adapter getUserMedia() | constraints: +1ms Object
voip.js?v10:25 callUp
voip.js?v10:764 UA
jssip-0.6.18.js:21376 rtcninja:Adapter getUserMedia() | success +1s
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection addStream() | stream: +1ms MediaStream
jssip-0.6.18.js:21376 JsSIP:RTCSession session connecting +0ms
voip.js?v10:268 message Connecting with user...
voip.js?v10:25 outgoingCall.connect
jssip-0.6.18.js:21376 JsSIP:RTCSession createLocalDescription() +5ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection createOffer() +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onnegotiationneeded() +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection createOffer() | success +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setLocalDescription() +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: have-local-offer +4ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setLocalDescription() | success +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:3859917557 1 udp 2122260223 192.168.1.96 49040 typ host generation 0 +12ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:3859917557 2 udp 2122260223 192.168.1.96 49040 typ host generation 0 +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:2828162565 1 tcp 1518280447 192.168.1.96 0 typ host tcptype active generation 0 +89ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:2828162565 2 tcp 1518280447 192.168.1.96 0 typ host tcptype active generation 0 +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:268555041 1 udp 1686052607 178.219.174.6 49040 typ srflx raddr 192.168.1.96 rport 49040 generation 0 +15ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:268555041 2 udp 1686052607 178.219.174.6 49040 typ srflx raddr 192.168.1.96 rport 49040 generation 0 +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onicecandidate() | end of candidates +34ms
jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

INVITE sip:888@54.200.180.31 SIP/2.0
Via: SIP/2.0/WS 192.0.2.190;branch=z9hG4bK3369694
Max-Forwards: 69
To: <sip:888@54.200.180.31>
From: "777" <sip:777@54.200.180.31>;tag=runhooterc
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 5786 INVITE
Contact: <sip:uvqkt7q5@192.0.2.190;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: ice,outbound
User-Agent: JsSIP 0.6.18
Content-Length: 1755

v=0
o=- 5379936291811845382 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS qS2aF1xV8mHW0295EerOolXjVOEjIuFdbxTl
m=audio 49040 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 178.219.174.6
a=rtcp:49040 IN IP4 178.219.174.6
a=candidate:3859917557 1 udp 2122260223 192.168.1.96 49040 typ host generation 0
a=candidate:3859917557 2 udp 2122260223 192.168.1.96 49040 typ host generation 0
a=candidate:2828162565 1 tcp 1518280447 192.168.1.96 0 typ host tcptype active generation 0
a=candidate:2828162565 2 tcp 1518280447 192.168.1.96 0 typ host tcptype active generation 0
a=candidate:268555041 1 udp 1686052607 178.219.174.6 49040 typ srflx raddr 192.168.1.96 rport 49040 generation 0
a=candidate:268555041 2 udp 1686052607 178.219.174.6 49040 typ srflx raddr 192.168.1.96 rport 49040 generation 0
a=ice-ufrag:byVbgYD7ht+TtMDu
a=ice-pwd:P58JbmbRDLcAW6CEa8XmP3Wm
a=ice-options:google-ice
a=fingerprint:sha-256 9A:C1:2D:20:4C:16:6A:C7:A4:17:06:E8:02:14:48:25:EA:A3:37:5D:F4:34:E9:76:2C:83:75:C2:C5:63:5A:0B
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2100607185 cname:a9WADR7qZ7mR7TFe
a=ssrc:2100607185 msid:qS2aF1xV8mHW0295EerOolXjVOEjIuFdbxTl 37ea7d5d-1c9d-4e08-b619-b9245a60230c
a=ssrc:2100607185 mslabel:qS2aF1xV8mHW0295EerOolXjVOEjIuFdbxTl
a=ssrc:2100607185 label:37ea7d5d-1c9d-4e08-b619-b9245a60230c

 +5ms
jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 192.0.2.190;branch=z9hG4bK3369694;received=178.219.174.6;rport=42912
From: "777" <sip:777@54.200.180.31>;tag=runhooterc
To: <sip:888@54.200.180.31>;tag=as4b0121bd
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 5786 INVITE
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="54.200.180.31", nonce="46a1b05e"
Content-Length: 0


 +233ms
jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

ACK sip:888@54.200.180.31 SIP/2.0
Via: SIP/2.0/WS 192.0.2.190;branch=z9hG4bK3369694
To: <sip:888@54.200.180.31>;tag=as4b0121bd
From: "777" <sip:777@54.200.180.31>;tag=runhooterc
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 5786 ACK
Content-Length: 0


 +17ms
jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

INVITE sip:888@54.200.180.31 SIP/2.0
Via: SIP/2.0/WS 192.0.2.190;branch=z9hG4bK890423
Max-Forwards: 69
To: <sip:888@54.200.180.31>
From: "777" <sip:777@54.200.180.31>;tag=runhooterc
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 5787 INVITE
Authorization: Digest algorithm=MD5, username="777", realm="54.200.180.31", nonce="46a1b05e", uri="sip:888@54.200.180.31", response="35f0428fbaa480f4c3f108b35df6bc1f"
Contact: <sip:uvqkt7q5@192.0.2.190;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: ice,outbound
User-Agent: JsSIP 0.6.18
Content-Length: 1755

v=0
o=- 5379936291811845382 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS qS2aF1xV8mHW0295EerOolXjVOEjIuFdbxTl
m=audio 49040 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 178.219.174.6
a=rtcp:49040 IN IP4 178.219.174.6
a=candidate:3859917557 1 udp 2122260223 192.168.1.96 49040 typ host generation 0
a=candidate:3859917557 2 udp 2122260223 192.168.1.96 49040 typ host generation 0
a=candidate:2828162565 1 tcp 1518280447 192.168.1.96 0 typ host tcptype active generation 0
a=candidate:2828162565 2 tcp 1518280447 192.168.1.96 0 typ host tcptype active generation 0
a=candidate:268555041 1 udp 1686052607 178.219.174.6 49040 typ srflx raddr 192.168.1.96 rport 49040 generation 0
a=candidate:268555041 2 udp 1686052607 178.219.174.6 49040 typ srflx raddr 192.168.1.96 rport 49040 generation 0
a=ice-ufrag:byVbgYD7ht+TtMDu
a=ice-pwd:P58JbmbRDLcAW6CEa8XmP3Wm
a=ice-options:google-ice
a=fingerprint:sha-256 9A:C1:2D:20:4C:16:6A:C7:A4:17:06:E8:02:14:48:25:EA:A3:37:5D:F4:34:E9:76:2C:83:75:C2:C5:63:5A:0B
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:126 telephone-event/8000
a=maxptime:60
a=ssrc:2100607185 cname:a9WADR7qZ7mR7TFe
a=ssrc:2100607185 msid:qS2aF1xV8mHW0295EerOolXjVOEjIuFdbxTl 37ea7d5d-1c9d-4e08-b619-b9245a60230c
a=ssrc:2100607185 mslabel:qS2aF1xV8mHW0295EerOolXjVOEjIuFdbxTl
a=ssrc:2100607185 label:37ea7d5d-1c9d-4e08-b619-b9245a60230c

 +5ms
jssip-0.6.18.js:21376 JsSIP:InviteClientTransaction Timer D expired for transaction z9hG4bK3369694 +0ms
jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS 192.0.2.190;branch=z9hG4bK890423;received=178.219.174.6;rport=42912
From: "777" <sip:777@54.200.180.31>;tag=runhooterc
To: <sip:888@54.200.180.31>
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 5787 INVITE
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:888@54.200.180.31:0;transport=WS>
Content-Length: 0


 +484ms
jssip-0.6.18.js:21376 JsSIP:RTCSession receiveInviteResponse() +10ms
jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.0.2.190;branch=z9hG4bK890423;received=178.219.174.6;rport=42912
From: "777" <sip:777@54.200.180.31>;tag=runhooterc
To: <sip:888@54.200.180.31>;tag=as0278aeb5
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 5787 INVITE
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:888@54.200.180.31:0;transport=WS>
Content-Length: 0


 +117ms
jssip-0.6.18.js:21376 JsSIP:RTCSession receiveInviteResponse() +6ms
jssip-0.6.18.js:21376 JsSIP:Dialog new UAC dialog created with status EARLY +2ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session progress +0ms
voip.js?v10:262 debug call progress Object
voip.js?v10:268 message Calling...
voip.js?v10:25 outgoingCall.wait
jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.0.2.190;branch=z9hG4bK890423;received=178.219.174.6;rport=42912
From: "777" <sip:777@54.200.180.31>;tag=runhooterc
To: <sip:888@54.200.180.31>;tag=as0278aeb5
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 5787 INVITE
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:888@54.200.180.31:0;transport=WS>
Content-Length: 0


 +251ms
jssip-0.6.18.js:21376 JsSIP:RTCSession receiveInviteResponse() +6ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session progress +1ms
voip.js?v10:262 debug call progress Object
voip.js?v10:268 message Calling...
voip.js?v10:25 outgoingCall.wait
jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.190;branch=z9hG4bK890423;received=178.219.174.6;rport=42912
From: "777" <sip:777@54.200.180.31>;tag=runhooterc
To: <sip:888@54.200.180.31>;tag=as0278aeb5
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 5787 INVITE
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:888@54.200.180.31:0;transport=WS>
Content-Type: application/sdp
Content-Length: 808

v=0
o=root 204618965 204618965 IN IP4 54.200.180.31
s=Asterisk PBX 11.13.1
c=IN IP4 54.200.180.31
t=0 0
m=audio 13664 RTP/SAVPF 0 8 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:4333a3aa4329771746ddd5a504a6d30b
a=ice-pwd:7fc0a5a868a3f4b842d75dca216b1e10
a=candidate:Hac1f0f11 1 UDP 2130706431 172.31.15.17 13664 typ host
a=candidate:S36c8b41f 1 UDP 1694498815 54.200.180.31 13664 typ srflx
a=candidate:Hac1f0f11 2 UDP 2130706430 172.31.15.17 13665 typ host
a=candidate:S36c8b41f 2 UDP 1694498814 54.200.180.31 13665 typ srflx
a=connection:new
a=setup:active
a=fingerprint:SHA-256 34:1F:33:1C:B5:14:3D:D9:F4:25:7E:29:60:96:8A:61:53:F0:B9:B4:47:1E:A5:E4:5A:82:DC:A5:1E:FF:1E:94
a=sendrecv

 +4s
jssip-0.6.18.js:21376 JsSIP:RTCSession receiveInviteResponse() +4ms
jssip-0.6.18.js:21376 JsSIP:Dialog dialog vm16crlhg9d5g8d56tr6runhootercas0278aeb5  changed to CONFIRMED state +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setRemoteDescription() +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: stable +3ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onaddstream() | stream: +1ms MediaStream
voip.js?v10:262 debug addstream Object
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection oniceconnectionstatechange() | iceConnectionState: checking +1ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection setRemoteDescription() | success +0ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session accepted +0ms
voip.js?v10:262 debug call accepted Object
jssip-0.6.18.js:21376 JsSIP:RTCSession sendRequest() +1ms
jssip-0.6.18.js:21376 JsSIP:RTCSession:Request new | ACK +0ms
jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

ACK sip:888@54.200.180.31:0;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.0.2.190;branch=z9hG4bK101623
Max-Forwards: 69
To: <sip:888@54.200.180.31>;tag=as0278aeb5
From: "777" <sip:777@54.200.180.31>;tag=runhooterc
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 5787 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: outbound
User-Agent: JsSIP 0.6.18
Content-Length: 0


 +2ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session confirmed +0ms
voip.js?v10:262 debug call confirmed Object
voip.js?v10:268 message Speaking...
voip.js?v10:25 outgoingCall.speak
voip.js?v10:712 call confirmed
voip.js?v10:268 00:00:01
voip.js?v10:268 00:00:02
voip.js?v10:268 00:00:03
voip.js?v10:268 00:00:04
voip.js?v10:268 00:00:05
voip.js?v10:268 00:00:06
voip.js?v10:268 00:00:07
voip.js?v10:268 00:00:08
voip.js?v10:268 00:00:09
voip.js?v10:268 00:00:10
jssip-0.6.18.js:21376 JsSIP:Transport received WebSocket text message:

BYE sip:uvqkt7q5@192.0.2.190;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 54.200.180.31:0;branch=z9hG4bK1ebe6a01;rport
Max-Forwards: 70
From: <sip:888@54.200.180.31>;tag=as0278aeb5
To: "777" <sip:777@54.200.180.31>;tag=runhooterc
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.13.1
Proxy-Authorization: Digest username="777", realm="54.200.180.31", algorithm=MD5, uri="sip:54.200.180.31", nonce="46a1b05e", response="86d9aff55bcae52746307cd3542b6184"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


 +11s
jssip-0.6.18.js:21376 JsSIP:RTCSession receiveRequest() +12ms
jssip-0.6.18.js:21376 JsSIP:Transport sending WebSocket message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 54.200.180.31:0;branch=z9hG4bK1ebe6a01;rport
To: "777" <sip:777@54.200.180.31>;tag=runhooterc
From: <sip:888@54.200.180.31>;tag=as0278aeb5
Call-ID: vm16crlhg9d5g8d56tr6
CSeq: 102 BYE
Supported: outbound
Content-Length: 0


 +1ms
jssip-0.6.18.js:21376 JsSIP:RTCSession session ended +0ms
jssip-0.6.18.js:21376 JsSIP:RTCSession close() +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection close() +0ms
jssip-0.6.18.js:21376 JsSIP:RTCSession close() | closing local MediaStream +2ms
jssip-0.6.18.js:21376 rtcninja:Adapter closeMediaStream() | calling stop() on all the MediaStreamTrack +0ms
jssip-0.6.18.js:21376 JsSIP:Dialog dialog vm16crlhg9d5g8d56tr6runhootercas0278aeb5 deleted +0ms
voip.js?v10:262 debug call ended Object
voip.js?v10:268 message Call ended
voip.js?v10:25 outgoingCall.end
voip.js?v10:268 Call ended, duration 00:00:10
jssip-0.6.18.js:21376 JsSIP:NonInviteServerTransaction Timer J expired for transaction z9hG4bK1ebe6a01 +2ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection oniceconnectionstatechange() | iceConnectionState: closed +0ms
jssip-0.6.18.js:21376 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: closed +0ms
jssip-0.6.18.js:21376 JsSIP:InviteClientTransaction Timer B expired for transaction z9hG4bK890423 +16s
jssip-0.6.18.js:21376 JsSIP:InviteClientTransaction Timer M expired for transaction z9hG4bK890423 +5s

Thanks for any help.

genobe · February 17, 2015, 6:03pm

I’m no help to you, but I’ve got the same problem (viewtopic.php?f=1&t=92508).

Just putting this out here in case this is of use to anyone with experience (or maybe either of us finds a solution and we can help eachother out!).

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Asterisk + JsSIP + Amazon EC2

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