DoubangoTelecom / sipml5

The world's first HTML5 SIP client (WebRTC)
BSD 3-Clause "New" or "Revised" License
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how to avoid 603 decline #358

Open lcomao opened 2 years ago

lcomao commented 2 years ago

when i make call to web, sometimes web can not accept and return 603 decline, log like below:

layouts__BasicLayout.df33f3ae.async.js:1 >>>sip stack events listener Event: i_new_call sip.min.js:1 State machine: s0000_Ringing_2_Terminated_X_Reject sip.min.js:1 === INVITE Dialog terminated === sip.min.js:1 PeerConnection::stop() sip.min.js:1 State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699 sip.min.js:1 SEND: SIP/2.0 603 Decline Via: SIP/2.0/WSS 47.99.185.209:47060;rport=47060;branch=z9hG4bK2UgX82Ncv2rpQ From: "17316375517"sip:17316375517@47.99.185.209;tag=190jKa55ZvyNg To: sip:16217500003@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss;tag=BnbuPTBGVQ2y6TgQyEsL Call-ID: 6378b4c1-8bd8-123a-45a4-00163f004e49 CSeq: 41030319 INVITE Content-Length: 0 Reason: SIP; cause=603; text="Decline"