{"action":"opened","issue":{"url":"https://api.github.com/repos/DoubangoTelecom/webrtc2sip/issues/187","labels_url":"https://api.github.com/repos/DoubangoTelecom/webrtc2sip/issues/187/labels{/name}","comments_url":"https://api.github.com/repos/DoubangoTelecom/webrtc2sip/issues/187/comments","events_url":"https://api.github.com/repos/DoubangoTelecom/webrtc2sip/issues/187/events","html_url":"https://github.com/DoubangoTelecom/webrtc2sip/issues/187","id":112718446,"number":187,"title":"No Audio if call is accepted after 30 seconds ","user":{"login":"vinodpandey23","id":15241330,"avatar_url":"https://avatars.githubusercontent.com/u/15241330?v=3","gravatar_id":"","url":"https://api.github.com/users/vinodpandey23","html_url":"https://github.com/vinodpandey23","followers_url":"https://api.github.com/users/vinodpandey23/followers","following_url":"https://api.github.com/users/vinodpandey23/following{/other_user}","gists_url":"https://api.github.com/users/vinodpandey23/gists{/gist_id}","starred_url":"https://api.github.com/users/vinodpandey23/starred{/owner}{/repo}","subscriptions_url":"https://api.github.com/users/vinodpandey23/subscriptions","organizations_url":"https://api.github.com/users/vinodpandey23/orgs","repos_url":"https://api.github.com/users/vinodpandey23/repos","events_url":"https://api.github.com/users/vinodpandey23/events{/privacy}","received_events_url":"https://api.github.com/users/vinodpandey23/received_events","type":"User","site_admin":false},"labels":[],"state":"open","locked":false,"assignee":null,"milestone":null,"comments":0,"created_at":"2015-10-22T02:26:51Z","updated_at":"2015-10-22T02:26:51Z","closed_at":null,"body":"I am using Asterisk 11.6-cert9 with webrtc2sip and sipml5 javascript library for voice over web.\r\n\r\nCall is initiated from legacy SIP client and received from sipml5 supported web page. Call is connected and successful audio at both end. \r\n\r\nHowever if call is received after waiting 30 seconds or later of notification; it is connected successful but no audio at both end. I can see below line in webrtc2sip logs:\r\n\"Audio producer is not yet started\""}}