{"action":"created","issue":{"url":"https://api.github.com/repos/onsip/SIP.js/issues/257","labels_url":"https://api.github.com/repos/onsip/SIP.js/issues/257/labels{/name}","comments_url":"https://api.github.com/repos/onsip/SIP.js/issues/257/comments","events_url":"https://api.github.com/repos/onsip/SIP.js/issues/257/events","html_url":"https://github.com/onsip/SIP.js/issues/257","id":121026047,"number":257,"title":"Trouble answering call in Chrome (47), works fine in FF (42)","user":{"login":"effeks","id":12557739,"avatar_url":"https://avatars.githubusercontent.com/u/12557739?v=3","gravatar_id":"","url":"https://api.github.com/users/effeks","html_url":"https://github.com/effeks","followers_url":"https://api.github.com/users/effeks/followers","following_url":"https://api.github.com/users/effeks/following{/other_user}","gists_url":"https://api.github.com/users/effeks/gists{/gist_id}","starred_url":"https://api.github.com/users/effeks/starred{/owner}{/repo}","subscriptions_url":"https://api.github.com/users/effeks/subscriptions","organizations_url":"https://api.github.com/users/effeks/orgs","repos_url":"https://api.github.com/users/effeks/repos","events_url":"https://api.github.com/users/effeks/events{/privacy}","received_events_url":"https://api.github.com/users/effeks/received_events","type":"User","site_admin":false},"labels":[],"state":"open","locked":false,"assignee":null,"milestone":null,"comments":10,"created_at":"2015-12-08T15:05:54Z","updated_at":"2016-01-16T23:40:36Z","closed_at":null,"body":"Hi!\r\n\r\nI'm trying to accept an incoming call from a SIP phone in a web browser (via a Kamailio/RTPEngine setup). My code works fine in Firefox (42) where the call is accepted and the two endpoints can talk without problem. \r\n\r\nIn Chrome (47) however, the last message sent out is \"180 Ringing\" and Chrome never sends the \"200 OK\". If I look into \"chrome://webrtc-internals/\" it seems the response (or at least part of the response) is created but again, never sent out.\r\n\r\nThese are the last messages in the console:\r\n```\r\nTue Dec 08 2015 15:43:18 GMT+0100 (CET) | sip.invitecontext.mediahandler | acquiring local media sip-0.7.2.js:2892\r\nTue Dec 08 2015 15:43:18 GMT+0100 (CET) | sip.invitecontext.mediahandler | stream added: default sip-0.7.2.js:2892\r\nTue Dec 08 2015 15:43:18 GMT+0100 (CET) | sip.invitecontext.mediahandler | acquired local media streams\r\n```\r\nafter this nothing more happens and the caller just hears ringtones until timeout.\r\n\r\nSame log from Firefox:\r\n```\r\nTue Dec 08 2015 14:12:32 GMT+0100 (CET) | sip.invitecontext.mediahandler | acquiring local media sip-0.7.2.js:2892:3\r\nTue Dec 08 2015 14:12:35 GMT+0100 (CET) | sip.invitecontext.mediahandler | acquired local media streams sip-0.7.2.js:2892:3\r\nTue Dec 08 2015 14:12:35 GMT+0100 (CET) | sip.invitecontext.mediahandler | ICE candidate received: candidate:0 1 UDP 2130379007 my_ip_here 63981 typ host sip-0.7.2.js:2892:3\r\nTue Dec 08 2015 14:12:35 GMT+0100 (CET) | sip.transport | sending WebSocket message:\r\n```\r\n\r\nI found this thread http://stackoverflow.com/questions/27489881/webrtc-never-fires-onicecandidate which points at the OfferToReceiveAudio option but changing that seems to make no difference what so ever."},"comment":{"url":"https://api.github.com/repos/onsip/SIP.js/issues/comments/172274145","html_url":"https://github.com/onsip/SIP.js/issues/257#issuecomment-172274145","issue_url":"https://api.github.com/repos/onsip/SIP.js/issues/257","id":172274145,"user":{"login":"bthai","id":2927659,"avatar_url":"https://avatars.githubusercontent.com/u/2927659?v=3","gravatar_id":"","url":"https://api.github.com/users/bthai","html_url":"https://github.com/bthai","followers_url":"https://api.github.com/users/bthai/followers","following_url":"https://api.github.com/users/bthai/following{/other_user}","gists_url":"https://api.github.com/users/bthai/gists{/gist_id}","starred_url":"https://api.github.com/users/bthai/starred{/owner}{/repo}","subscriptions_url":"https://api.github.com/users/bthai/subscriptions","organizations_url":"https://api.github.com/users/bthai/orgs","repos_url":"https://api.github.com/users/bthai/repos","events_url":"https://api.github.com/users/bthai/events{/privacy}","received_events_url":"https://api.github.com/users/bthai/received_events","type":"User","site_admin":false},"created_at":"2016-01-16T23:40:36Z","updated_at":"2016-01-16T23:40:36Z","body":"looks like if i take the wss url and change it to https and type that into chrome and accept the warning message and try connecting again, i can connect\r\n```\r\nhttps://192.168.10.145:8089/asterisk/ws\r\n```\r\n\r\nonce i do that, i'm pretty much back to my original issue. FF 43 works, i can accept/make calls but on chrome 47, i can't and it never prompts me for what audio i want to use..."}}