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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 #265

Closed mitesh1980 closed 8 years ago

mitesh1980 commented 8 years ago

Hi @egreenmachine,

I am trying to call using Asterisk 11.11 only, not using FreeSwitch now.

I am successfull to get audio call using 3cx softphone on calling 1061 (Legacy Client) to 1062 (Legacy Client)

Problem: When i try t call using sip.js and WebRTC client (1060) to 1061 (Legacy Client). call get connected but no audio.

Asterisk log:

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 icesupport=yes ;

[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:

[general] [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 log ::

Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | configuration parameters after validation: sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · viaHost: "192.0.2.39" sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · uri: sip:1060@192.168.55.55 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 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) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · password: NOT SHOWN sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · registerExpires: 600 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · register: true sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · registrarServer: sip:192.168.55.55 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · wsServerMaxReconnection: 3 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · wsServerReconnectionTimeout: 10 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · connectionRecoveryMinInterval: 2 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · connectionRecoveryMaxInterval: 30 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · keepAliveInterval: 0 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · usePreloadedRoute: false sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · userAgentString: "SIP.js/0.7.1" sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · iceCheckingTimeout: 5000 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · noAnswerTimeout: 60000 sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · stunServers: ["stun:null"] sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · turnServers: [] sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · traceSip: true sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · hackViaTcp: false sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · hackIpInContact: true sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · hackWssInTransport: false sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · contactTransport: "ws" sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · forceRport: false sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · autostart: true sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · rel100: "none" sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · replaces: "none" sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:47 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) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · authenticationFactory: undefined sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · authorizationUser: "1060" sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · instanceId: "1cf105ad-f426-4279-ac8e-d0d9710fafe6" sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · sipjsId: "lmfem" sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · hostportParams: "192.168.55.55" sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | · media: undefined sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.ua | user requested startup... sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:47 GMT+0530 (India Standard Time) | sip.transport | connecting to WebSocket ws://192.168.55.55:8088/ws sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:48 GMT+0530 (India Standard Time) | sip.transport | WebSocket ws://192.168.55.55:8088/ws connected sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:48 GMT+0530 (India Standard Time) | sip.ua | connection state set to 0 sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:48 GMT+0530 (India Standard Time) | sip.transport | sending WebSocket message:

REGISTER sip:192.168.55.55 SIP/2.0 Via: SIP/2.0/WS 192.0.2.39;branch=z9hG4bK1439709 Max-Forwards: 70 To: sip:1060@192.168.55.55 From: sip:1060@192.168.55.55;tag=c3fc38pv8r Call-ID: o85ri2j93p9m5gjupdedhq CSeq: 81 REGISTER Contact: sip:pmh91k37@192.0.2.39;transport=ws;reg-id=1;+sip.instance="urn:uuid:1cf105ad-f426-4279-ac8e-d0d9710fafe6";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)

Thu Dec 24 2015 16:41:49 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 192.0.2.39;branch=z9hG4bK1439709;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=c3fc38pv8r To: sip:1060@192.168.55.55;tag=as0cb422fb Call-ID: o85ri2j93p9m5gjupdedhq 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="3d0bad12" Content-Length: 0

sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:49 GMT+0530 (India Standard Time) | sip.transport | sending WebSocket message:

REGISTER sip:192.168.55.55 SIP/2.0 Via: SIP/2.0/WS 192.0.2.39;branch=z9hG4bK5260782 Max-Forwards: 70 To: sip:1060@192.168.55.55 From: sip:1060@192.168.55.55;tag=c3fc38pv8r Call-ID: o85ri2j93p9m5gjupdedhq CSeq: 82 REGISTER Authorization: Digest algorithm=MD5, username="1060", realm="192.168.55.55", nonce="3d0bad12", uri="sip:192.168.55.55", response="9cf40844dd09cb8541d89344620c4481" Contact: sip:pmh91k37@192.0.2.39;transport=ws;reg-id=1;+sip.instance="urn:uuid:1cf105ad-f426-4279-ac8e-d0d9710fafe6";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)

Thu Dec 24 2015 16:41:49 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 200 OK Via: SIP/2.0/WS 192.0.2.39;branch=z9hG4bK5260782;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=c3fc38pv8r To: sip:1060@192.168.55.55;tag=as0cb422fb Call-ID: o85ri2j93p9m5gjupdedhq 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:pmh91k37@192.0.2.39;transport=ws;expires=600 Date: Thu, 24 Dec 2015 03:11:38 GMT Content-Length: 0

sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:50 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | acquiring local media sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:51 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | acquired local media streams sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:51 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | ICE candidate received: candidate:0 1 UDP 2128609535 192.168.5.66 50458 typ host sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:51 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | ICE candidate received: candidate:0 2 UDP 2128609534 192.168.5.66 50459 typ host sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:53 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 192.0.2.39;branch=z9hG4bK9277042 Max-Forwards: 70 To: sip:1061@192.168.55.55 From: sip:1060@192.168.55.55;tag=e0nc9etqao Call-ID: lmfemhosah81njiijop6 CSeq: 7555 INVITE Contact: sip:pmh91k37@192.0.2.39;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: 916

v=0 o=mozilla...THIS_IS_SDPARTA-39.0 4294967295 0 IN IP4 192.0.2.67 s=- t=0 0 a=sendrecv a=fingerprint:sha-256 E9:19:C3:31:90:BB:11:56:15:E1:CE:8B:92:03:EB:73:2F:C9:47:E0:52:95:2B:BF:2B:B4:A6:C7:10:23:B5:C1 a=group:BUNDLE sdparta_0 a=ice-options:trickle a=msid-semantic:WMS * m=audio 50458 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 192.168.5.66 a=candidate:0 1 UDP 2128609535 192.168.5.66 50458 typ host a=candidate:0 2 UDP 2128609534 192.168.5.66 50459 typ host a=sendrecv a=end-of-candidates a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=ice-pwd:0e1097b0ff55158a8feeed2696e32d0e a=ice-ufrag:8c3ffa52 a=mid:sdparta_0 a=msid:{7d1f65c7-21f4-4528-95e0-21e9262c7fde} {844c2c22-3ead-407d-9b51-37aa45d4c836} 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:522719446 cname:{0afcc099-72d5-4ae2-a837-ca09e9079b0e}

sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:53 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 192.0.2.39;branch=z9hG4bK9277042;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=e0nc9etqao To: sip:1061@192.168.55.55;tag=as15dbc943 Call-ID: lmfemhosah81njiijop6 CSeq: 7555 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="57c93e71" Content-Length: 0

sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:53 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 192.0.2.39;branch=z9hG4bK9277042 To: sip:1061@192.168.55.55;tag=as15dbc943 From: sip:1060@192.168.55.55;tag=e0nc9etqao Call-ID: lmfemhosah81njiijop6 CSeq: 7555 ACK

sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:53 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 192.0.2.39;branch=z9hG4bK2032100 Max-Forwards: 70 To: sip:1061@192.168.55.55 From: sip:1060@192.168.55.55;tag=e0nc9etqao Call-ID: lmfemhosah81njiijop6 CSeq: 7556 INVITE Authorization: Digest algorithm=MD5, username="1060", realm="192.168.55.55", nonce="57c93e71", uri="sip:1061@192.168.55.55", response="556800dab595839a7ad76e86f7c4d65c" Contact: sip:pmh91k37@192.0.2.39;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: 916

v=0 o=mozilla...THIS_IS_SDPARTA-39.0 4294967295 0 IN IP4 192.0.2.67 s=- t=0 0 a=sendrecv a=fingerprint:sha-256 E9:19:C3:31:90:BB:11:56:15:E1:CE:8B:92:03:EB:73:2F:C9:47:E0:52:95:2B:BF:2B:B4:A6:C7:10:23:B5:C1 a=group:BUNDLE sdparta_0 a=ice-options:trickle a=msid-semantic:WMS * m=audio 50458 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 192.168.5.66 a=candidate:0 1 UDP 2128609535 192.168.5.66 50458 typ host a=candidate:0 2 UDP 2128609534 192.168.5.66 50459 typ host a=sendrecv a=end-of-candidates a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=ice-pwd:0e1097b0ff55158a8feeed2696e32d0e a=ice-ufrag:8c3ffa52 a=mid:sdparta_0 a=msid:{7d1f65c7-21f4-4528-95e0-21e9262c7fde} {844c2c22-3ead-407d-9b51-37aa45d4c836} 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:522719446 cname:{0afcc099-72d5-4ae2-a837-ca09e9079b0e}

sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:53 GMT+0530 (India Standard Time) | sip.transaction.ict | Timer D expired for INVITE client transaction z9hG4bK9277042 sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:53 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 100 Trying Via: SIP/2.0/WS 192.0.2.39;branch=z9hG4bK2032100;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=e0nc9etqao To: sip:1061@192.168.55.55 Call-ID: lmfemhosah81njiijop6 CSeq: 7556 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)

Thu Dec 24 2015 16:41:54 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.0.2.39;branch=z9hG4bK2032100;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=e0nc9etqao To: sip:1061@192.168.55.55;tag=as0184df39 Call-ID: lmfemhosah81njiijop6 CSeq: 7556 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) Thu Dec 24 2015 16:41:54 GMT+0530 (India Standard Time) | sip.dialog | new UAC dialog created with status EARLY sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:56 GMT+0530 (India Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 200 OK Via: SIP/2.0/WS 192.0.2.39;branch=z9hG4bK2032100;received=192.168.5.66 From: sip:1060@192.168.55.55;tag=e0nc9etqao To: sip:1061@192.168.55.55;tag=as0184df39 Call-ID: lmfemhosah81njiijop6 CSeq: 7556 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 2140861074 2140861074 IN IP4 192.168.55.55 s=Asterisk PBX 11.11.0 c=IN IP4 192.168.55.55 t=0 0 m=audio 16732 RTP/SAVPF 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=ice-ufrag:47746d173c7099c4484c547e25a58535 a=ice-pwd:4ffd9a360a9d1df53da80951359675c9 a=candidate:Hc0a83737 1 UDP 2130706431 192.168.55.55 16732 typ host a=candidate:Hc0a83737 2 UDP 2130706430 192.168.55.55 16733 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)

Thu Dec 24 2015 16:41:56 GMT+0530 (India Standard Time) | sip.dialog | dialog lmfemhosah81njiijop6e0nc9etqaoas0184df39 changed to CONFIRMED state

sip-0.7.1.js (line 2810)

Thu Dec 24 2015 16:41:56 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 192.0.2.39;branch=z9hG4bK4269294 Max-Forwards: 70 To: sip:1061@192.168.55.55;tag=as0184df39 From: sip:1060@192.168.55.55;tag=e0nc9etqao Call-ID: lmfemhosah81njiijop6 CSeq: 7556 ACK Supported: outbound User-Agent: SIP.js/0.7.1 Content-Length: 0

sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:41:56 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | stream added: {06f8cc61-c5bb-480e-9b70-66ab4749be6e} sip-0.7.1.js (line 2810) Thu Dec 24 2015 16:42:01 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | RTCIceChecking Timeout Triggered after 5000 milliseconds

Any help is highly appreciated

egreenmachine commented 8 years ago

Hi @mitesh1980,

This is not a SIP.js issue, but an Asterisk configuration issue. As I have stated in #264, for help with configuring Asterisk, FreeSWITCH, or any other PBX please start a discussion on our mailing list. We would like to keep Github issues for actual bugs within SIP.js.

Thanks