onsip / SIP.js

A simple, intuitive, and powerful JavaScript signaling library
https://sipjs.com
MIT License
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No Audio call on Peer to Peer connection #264

Closed NishatSharma closed 8 years ago

NishatSharma commented 8 years ago

Hi, I am using Asterisk 11.11 and centOs 6.5 with the reference from below link: ** sip.js 0.7.1 *** http://sipjs.com/guides/server-configuration/asterisk/

I have followed the same steps as mentioned in above link


When i try to call using 3cx phone from 1061 (Legacy Client) to 1062 (Legacy Client), Call get connected and i clearly heard voice from the other end.



                                                          Problem:

When i i try to call from 1060 (WebRTC Client) to 1061 (Legacy Client) using sip.js

CALL GET CONNECTED BUT CAN'T HEAR VOICE FROM OTHER END.


Client apps: Firefox 39.0

Below are the Asterisk config files:

****** http.conf ***** [general] enabled=yes bindaddr=192.168.55.55 ; Replace this with your IP address bindport=8088 ; Replace this with the port you want to listen

***** sip.conf ***** [general] realm=192.168.55.55 ; Replace this with your IP address udpbindaddr=192.168.55.55 ; Replace this with your IP address transport=udp

[1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=Aa1060 ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF for this peer icesupport=yes ; Tell Asterisk to use ICE for this peer context=default ; Tell Asterisk which context to use when this peer is dialing directmedia=no ; Asterisk will relay media for this peer transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11 dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer dtlsverify=no ; Tell Asterisk to not verify your DTLS certs dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS

[1061] ; This will be the legacy SIP client type=friend username=1061 host=dynamic secret=Aa1061 context=default

[1062] ; This will be the legacy SIP client type=friend username=1062 host=dynamic secret=Aa1062 context=default

***** extensions.conf ***** [default] exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060 exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061 exten => 1062,1,Dial(SIP/1062) ; Dialing 1062 will call the SIP client registered to 1062


                                                       Browser logs

Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | configuration parameters after validation: sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · viaHost: "192.0.2.77" sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · uri: sip:1060@192.168.55.55 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · wsServers: [{"ws_uri":"ws://192.168.55.55:8088/ws","sip_uri":"sip:192.168.55.55:8088;transport=ws;lr","weight":0,"status":0,"scheme":"WS"}] sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · password: NOT SHOWN sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · registerExpires: 600 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · register: true sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · registrarServer: sip:192.168.55.55 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · wsServerMaxReconnection: 3 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · wsServerReconnectionTimeout: 10 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · connectionRecoveryMinInterval: 2 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · connectionRecoveryMaxInterval: 30 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · keepAliveInterval: 0 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · usePreloadedRoute: false sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · userAgentString: "SIP.js/0.7.1" sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · iceCheckingTimeout: 5000 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · noAnswerTimeout: 60000 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · stunServers: ["stun:stun.l.google.com:19302"] sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · turnServers: [] sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · traceSip: true sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · hackViaTcp: false sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · hackIpInContact: true sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · hackWssInTransport: false sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · contactTransport: "ws" sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · forceRport: false sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · autostart: true sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · rel100: "none" sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · replaces: "none" sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · mediaHandlerFactory: function defaultFactory(session, options) { "use strict";

return new MediaHandler(session, options); }

sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · authenticationFactory: undefined sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · authorizationUser: "1060" sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · instanceId: "3fd288e7-dbba-4f7f-af15-dac6c500fd7d" sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · sipjsId: "ppm4j" sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · hostportParams: "192.168.55.55" sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | · media: undefined sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.ua | user requested startup... sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:21 GMT+0530 (India Standard Time) | sip.transport | connecting to WebSocket ws://192.168.55.55:8088/ws sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:22 GMT+0530 (India Standard Time) | sip.transport | WebSocket ws://192.168.55.55:8088/ws connected sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:22 GMT+0530 (India Standard Time) | sip.ua | connection state set to 0 sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:22 GMT+0530 (India Standard Time) | sip.transport | sending WebSocket message:

REGISTER sip:192.168.55.55 SIP/2.0 Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK8417301 Max-Forwards: 70 To: sip:1060@192.168.55.55 From: sip:1060@192.168.55.55;tag=b4gfu699q5 Call-ID: a4nf3heukv1lu917fnafh9 CSeq: 81 REGISTER Contact: sip:45l4e1p5@192.0.2.77;transport=ws;reg-id=1;+sip.instance="urn:uuid:3fd288e7-dbba-4f7f-af15-dac6c500fd7d";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY Supported: path,gruu,outbound User-Agent: SIP.js/0.7.1 Content-Length: 0

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:23 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK8417301;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=b4gfu699q5 To: sip:1060@192.168.55.55;tag=as424c86bc Call-ID: a4nf3heukv1lu917fnafh9 CSeq: 81 REGISTER Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="192.168.55.55", nonce="4a9de4e5" Content-Length: 0

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:23 GMT+0530 (India Standard Time) | sip.transport | sending WebSocket message:

REGISTER sip:192.168.55.55 SIP/2.0 Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK618983 Max-Forwards: 70 To: sip:1060@192.168.55.55 From: sip:1060@192.168.55.55;tag=b4gfu699q5 Call-ID: a4nf3heukv1lu917fnafh9 CSeq: 82 REGISTER Authorization: Digest algorithm=MD5, username="1060", realm="192.168.55.55", nonce="4a9de4e5", uri="sip:192.168.55.55", response="867dff9ed67661d4961758538159bbde" Contact: sip:45l4e1p5@192.0.2.77;transport=ws;reg-id=1;+sip.instance="urn:uuid:3fd288e7-dbba-4f7f-af15-dac6c500fd7d";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY Supported: path,gruu,outbound User-Agent: SIP.js/0.7.1 Content-Length: 0

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:23 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 200 OK Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK618983;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=b4gfu699q5 To: sip:1060@192.168.55.55;tag=as424c86bc Call-ID: a4nf3heukv1lu917fnafh9 CSeq: 82 REGISTER Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: sip:45l4e1p5@192.0.2.77;transport=ws;expires=600 Date: Wed, 23 Dec 2015 04:45:13 GMT Content-Length: 0

sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:23 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | acquiring local media sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:25 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | acquired local media streams sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:25 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | ICE candidate received: candidate:0 1 UDP 2128609535 192.168.5.66 50437 typ host sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:25 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | ICE candidate received: candidate:0 2 UDP 2128609534 192.168.5.66 50438 typ host sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:25 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | ICE candidate received: candidate:1 1 UDP 1692467199 202.131.123.146 18881 typ srflx raddr 192.168.5.66 rport 50437 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:25 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | ICE candidate received: candidate:1 2 UDP 1692467198 202.131.123.146 65424 typ srflx raddr 192.168.5.66 rport 50438 sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:25 GMT+0530 (India Standard Time) | sip.transport | sending WebSocket message:

INVITE sip:1061@192.168.55.55 SIP/2.0 Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK2052471 Max-Forwards: 70 To: sip:1061@192.168.55.55 From: sip:1060@192.168.55.55;tag=bdei776rn9 Call-ID: ppm4jpj78hkmqk3571tp CSeq: 1391 INVITE Contact: sip:45l4e1p5@192.0.2.77;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY Content-Type: application/sdp Supported: outbound User-Agent: SIP.js/0.7.1 Content-Length: 1110

v=0 o=mozilla...THIS_IS_SDPARTA-39.0 4294967295 0 IN IP4 192.0.2.212 s=- t=0 0 a=sendrecv a=fingerprint:sha-256 FF:A2:8A:DC:0E:07:23:8F:44:92:BD:24:DA:AA:B3:5D:95:E3:53:EC:56:62:75:1F:BB:6F:0D:FC:7C:DB:D2:54 a=group:BUNDLE sdparta_0 a=ice-options:trickle a=msid-semantic:WMS * m=audio 18881 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 202.131.123.146 a=candidate:0 1 UDP 2128609535 192.168.5.66 50437 typ host a=candidate:0 2 UDP 2128609534 192.168.5.66 50438 typ host a=candidate:1 1 UDP 1692467199 202.131.123.146 18881 typ srflx raddr 192.168.5.66 rport 50437 a=candidate:1 2 UDP 1692467198 202.131.123.146 65424 typ srflx raddr 192.168.5.66 rport 50438 a=sendrecv a=end-of-candidates a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=ice-pwd:1aa4c4c6e82d4bd9b61bfc8955079111 a=ice-ufrag:5cf0fc5f a=mid:sdparta_0 a=msid:{4ffac880-bd3c-4e0f-9980-94db3a80ac83} {6e95f371-a363-4d4c-b5cc-5deb8bab00c2} 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:556552029 cname:{7d6564f4-c1b8-4f3b-b634-fdd758171eb0}

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:26 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK2052471;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=bdei776rn9 To: sip:1061@192.168.55.55;tag=as7db80572 Call-ID: ppm4jpj78hkmqk3571tp CSeq: 1391 INVITE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="192.168.55.55", nonce="5367a6b5" Content-Length: 0

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:26 GMT+0530 (India Standard Time) | sip.transport | sending WebSocket message:

ACK sip:1061@192.168.55.55 SIP/2.0 Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK2052471 To: sip:1061@192.168.55.55;tag=as7db80572 From: sip:1060@192.168.55.55;tag=bdei776rn9 Call-ID: ppm4jpj78hkmqk3571tp CSeq: 1391 ACK

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:26 GMT+0530 (India Standard Time) | sip.transport | sending WebSocket message:

INVITE sip:1061@192.168.55.55 SIP/2.0 Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK844225 Max-Forwards: 70 To: sip:1061@192.168.55.55 From: sip:1060@192.168.55.55;tag=bdei776rn9 Call-ID: ppm4jpj78hkmqk3571tp CSeq: 1392 INVITE Authorization: Digest algorithm=MD5, username="1060", realm="192.168.55.55", nonce="5367a6b5", uri="sip:1061@192.168.55.55", response="4381429e850997e71d2ced35c7b4c952" Contact: sip:45l4e1p5@192.0.2.77;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY Content-Type: application/sdp Supported: outbound User-Agent: SIP.js/0.7.1 Content-Length: 1110

v=0 o=mozilla...THIS_IS_SDPARTA-39.0 4294967295 0 IN IP4 192.0.2.212 s=- t=0 0 a=sendrecv a=fingerprint:sha-256 FF:A2:8A:DC:0E:07:23:8F:44:92:BD:24:DA:AA:B3:5D:95:E3:53:EC:56:62:75:1F:BB:6F:0D:FC:7C:DB:D2:54 a=group:BUNDLE sdparta_0 a=ice-options:trickle a=msid-semantic:WMS * m=audio 18881 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 202.131.123.146 a=candidate:0 1 UDP 2128609535 192.168.5.66 50437 typ host a=candidate:0 2 UDP 2128609534 192.168.5.66 50438 typ host a=candidate:1 1 UDP 1692467199 202.131.123.146 18881 typ srflx raddr 192.168.5.66 rport 50437 a=candidate:1 2 UDP 1692467198 202.131.123.146 65424 typ srflx raddr 192.168.5.66 rport 50438 a=sendrecv a=end-of-candidates a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=ice-pwd:1aa4c4c6e82d4bd9b61bfc8955079111 a=ice-ufrag:5cf0fc5f a=mid:sdparta_0 a=msid:{4ffac880-bd3c-4e0f-9980-94db3a80ac83} {6e95f371-a363-4d4c-b5cc-5deb8bab00c2} 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:556552029 cname:{7d6564f4-c1b8-4f3b-b634-fdd758171eb0}

sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:26 GMT+0530 (India Standard Time) | sip.transaction.ict | Timer D expired for INVITE client transaction z9hG4bK2052471 sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:26 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 100 Trying Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK844225;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=bdei776rn9 To: sip:1061@192.168.55.55 Call-ID: ppm4jpj78hkmqk3571tp CSeq: 1392 INVITE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:1061@192.168.55.55:5060;transport=WS Content-Length: 0

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:15:26 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK844225;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=bdei776rn9 To: sip:1061@192.168.55.55;tag=as64ad6ade Call-ID: ppm4jpj78hkmqk3571tp CSeq: 1392 INVITE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:1061@192.168.55.55:5060;transport=WS Content-Length: 0

sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:26 GMT+0530 (India Standard Time) | sip.dialog | new UAC dialog created with status EARLY sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:15:58 GMT+0530 (India Standard Time) | sip.transaction.ict | Timer B expired for INVITE client transaction z9hG4bK844225 sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:16:26 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK844225;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=bdei776rn9 To: sip:1061@192.168.55.55;tag=as64ad6ade Call-ID: ppm4jpj78hkmqk3571tp CSeq: 1392 INVITE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:1061@192.168.55.55:5060;transport=WS Content-Length: 0

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:17:06 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 200 OK Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK844225;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=bdei776rn9 To: sip:1061@192.168.55.55;tag=as64ad6ade Call-ID: ppm4jpj78hkmqk3571tp CSeq: 1392 INVITE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:1061@192.168.55.55:5060;transport=WS Content-Type: application/sdp Content-Length: 589

v=0 o=root 1876956366 1876956366 IN IP4 192.168.55.55 s=Asterisk PBX 11.11.0 c=IN IP4 192.168.55.55 t=0 0 m=audio 18762 RTP/SAVPF 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=ice-ufrag:7ebf6a451ef956bb0796a6f4317f0f3c a=ice-pwd:2f04e27b6f850b1d69e09c874575487a a=candidate:Hc0a83737 1 UDP 2130706431 192.168.55.55 18762 typ host a=candidate:Hc0a83737 2 UDP 2130706430 192.168.55.55 18763 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 26:93:4F:49:44:CA:08:56:52:79:E6:2D:D3:2A:86:7A:A7:75:CD:EE:CD:13:B6:71:74:02:23:31:73:BA:35:D8 a=sendrecv

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:17:06 GMT+0530 (India Standard Time) | sip.dialog | dialog ppm4jpj78hkmqk3571tpbdei776rn9as64ad6ade changed to CONFIRMED state

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:17:06 GMT+0530 (India Standard Time) | sip.transport | sending WebSocket message:

ACK sip:1061@192.168.55.55:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.0.2.77;branch=z9hG4bK8183418 Max-Forwards: 70 To: sip:1061@192.168.55.55;tag=as64ad6ade From: sip:1060@192.168.55.55;tag=bdei776rn9 Call-ID: ppm4jpj78hkmqk3571tp CSeq: 1392 ACK Supported: outbound User-Agent: SIP.js/0.7.1 Content-Length: 0

sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:17:06 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | stream added: {fa80a672-af98-4ab6-8a44-4900c6e3bc26} sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:17:11 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | RTCIceChecking Timeout Triggered after 5000 milliseconds sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:17:19 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

BYE sip:45l4e1p5@192.0.2.77;transport=ws;ob SIP/2.0 Via: SIP/2.0/WS 192.168.55.55:5060;branch=z9hG4bK0dde47e4 Max-Forwards: 70 From: sip:1061@192.168.55.55;tag=as64ad6ade To: sip:1060@192.168.55.55;tag=bdei776rn9 Call-ID: ppm4jpj78hkmqk3571tp CSeq: 102 BYE User-Agent: Asterisk PBX 11.11.0 Proxy-Authorization: Digest username="1060", realm="192.168.55.55", algorithm=MD5, uri="sip:192.168.55.55", nonce="5367a6b5", response="c727f52ceaf50974abd1821ba4f4db17" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

sip-0.7.1.js (line 2810)

Wed Dec 23 2015 18:17:19 GMT+0530 (India Standard Time) | sip.transport | sending WebSocket message:

SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.55.55:5060;branch=z9hG4bK0dde47e4 To: sip:1060@192.168.55.55;tag=bdei776rn9 From: sip:1061@192.168.55.55;tag=as64ad6ade Call-ID: ppm4jpj78hkmqk3571tp CSeq: 102 BYE Supported: outbound User-Agent: SIP.js/0.7.1 Content-Length: 0

sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:17:19 GMT+0530 (India Standard Time) | sip.inviteclientcontext | closing INVITE session ppm4jpj78hkmqk3571tp98ju01709m sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:17:19 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | closing PeerConnection sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:17:19 GMT+0530 (India Standard Time) | sip.dialog | dialog ppm4jpj78hkmqk3571tpbdei776rn9as64ad6ade deleted sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:17:19 GMT+0530 (India Standard Time) | sip.transaction.nist | Timer J expired for non-INVITE server transaction z9hG4bK0dde47e4 sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:17:25 GMT+0530 (India Standard Time) | sip.inviteclientcontext | Error: Attempted to send BYE in a terminated session.

target.call(console, content);

sip-0.7.1.js (line 2810) Wed Dec 23 2015 18:17:38 GMT+0530 (India Standard Time) | sip.transaction.ict | Timer M expired for INVITE client transaction z9hG4bK844225 sip-0.7.1.js (line 2810)

Any help is highly appreciated.

egreenmachine commented 8 years ago

It appears that Asterisk is sending a BYE, and the call is terminating. Since this appears to be an Asterisk issue, I am going to close this issue. For help with Asterisk, could you please follow up on our mailing list and include logs from Asterisk? Thanks