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DoubangoTelecom / sipml5

The world's first HTML5 SIP client (WebRTC)
BSD 3-Clause "New" or "Revised" License
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Call establised both side but no audio in call.what is this invalid means which i get in log? #284

Open suhani2911 opened 7 years ago

suhani2911 commented 7 years ago

I am using Asterisk certified/13.13-cert1 chrome version: 56.0 firefox version: 51.0

__tsip_transport_ws_onmessage tsk_utils.js:116:50 recv=SIP/2.0 200 OK

Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=51064;received=192.168.0.26;branch=z9hG4bKaCzQR28T5umCAbX1FiOXHCDaWR4D78JM

From: "7040"sip:7040@192.168.0.17;tag=xlyEPfB3Q165GZQpo26w

To: "7040"sip:7040@192.168.0.17;tag=as244c720b

Contact: sips:7040@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss;expires=200

Call-ID: a13d7080-7f33-91c7-4c36-81a4c0523169

CSeq: 4010 REGISTER

Expires: 200

Content-Length: 0

Server: Asterisk PBX certified/13.13-cert1

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE

Supported: replaces,timer

Date: 22 Feb 2017 12:15:45 GMT;22

tsk_utils.js:116:50 State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_utils.js:116:50 ==session event = sent_request tsk_utils.js:116:50 State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister tsk_utils.js:116:50 SEND: REGISTER sip:192.168.0.17 SIP/2.0

Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKvfrwn7w8OZxTxatyTT5le4M35HK8IqFQ;rport

From: "7040"sip:7040@192.168.0.17;tag=xlyEPfB3Q165GZQpo26w

To: "7040"sip:7040@192.168.0.17

Contact: "7040"sips:7040@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"

Call-ID: a13d7080-7f33-91c7-4c36-81a4c0523169

CSeq: 4011 REGISTER

Content-Length: 0

Max-Forwards: 70

User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04

Organization: Doubango Telecom

tsk_utils.js:116:50 __tsip_transport_ws_onmessage tsk_utils.js:116:50 recv=OPTIONS sips:7040@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss SIP/2.0

Via: SIP/2.0/WS 192.168.0.17:5060;rport;branch=z9hG4bK67fbbdbc

From: "asterisk"sip:asterisk@192.168.0.17;tag=as5b5bd3ad

To: <sips:7040@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>

Contact: sip:asterisk@192.168.0.17:5060;transport=WS

Call-ID: 61e457da1d08ba616746f07b02b3ee18@192.168.0.17:5060

CSeq: 102 OPTIONS

Content-Length: 0

Max-Forwards: 70

User-Agent: Asterisk PBX certified/13.13-cert1

Date: 22 Feb 2017 12:18:04 GMT;22

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE

Supported: replaces,timer

tsk_utils.js:116:50 Not implemented tsk_utils.js:128:51 SEND: SIP/2.0 405 Method Not Allowed

Via: SIP/2.0/WS 192.168.0.17:5060;rport=5060;branch=z9hG4bK67fbbdbc

From: "asterisk"sip:asterisk@192.168.0.17;tag=as5b5bd3ad

To: <sips:7040@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>

Call-ID: 61e457da1d08ba616746f07b02b3ee18@192.168.0.17:5060

CSeq: 102 OPTIONS

Content-Length: 0

tsk_utils.js:116:50 __tsip_transport_ws_onmessage tsk_utils.js:116:50 recv=SIP/2.0 200 OK

Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=51064;received=192.168.0.26;branch=z9hG4bKvfrwn7w8OZxTxatyTT5le4M35HK8IqFQ

From: "7040"sip:7040@192.168.0.17;tag=xlyEPfB3Q165GZQpo26w

To: "7040"sip:7040@192.168.0.17;tag=as3a3b1976

Contact: <sips:7040@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200

Call-ID: a13d7080-7f33-91c7-4c36-81a4c0523169

CSeq: 4011 REGISTER

Expires: 200

Content-Length: 0

Server: Asterisk PBX certified/13.13-cert1

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE

Supported: replaces,timer

Date: 22 Feb 2017 12:18:04 GMT;22

tsk_utils.js:116:50 State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx****