tag:blogger.com,1999:blog-7262335442574749724.post3250225123390837982..comments2023-09-24T21:45:15.090+08:00Comments on SANJAY WILLIE'S Human Language. Asterisk | Nagios | OpenSource | Microsoft | Security: OPUS codec with transcoding on Asterisk 11.5.x (or higher, 11.6,11.7,11.8,11.9) with(out) FreePBXJayWShttp://www.blogger.com/profile/04318296929423691109noreply@blogger.comBlogger1125tag:blogger.com,1999:blog-7262335442574749724.post-55812095122926137912014-06-03T14:07:06.977+08:002014-06-03T14:07:06.977+08:00Hi Sanjay,
Thanks for your excellent tutorials.
N...Hi Sanjay,<br /><br />Thanks for your excellent tutorials.<br />Now I'm getting struggled in asterisk with webRTC support.<br /><br />WebRTC with asterisk not working!<br /><br />I'm using your pre configured vm image disk for testing webRTC support. And also configured my own asterisk followed by your tutorials.<br />Chrome 34 version works for me.<br />Prior google chrome update,it works fine. but after chrome updated(stable), now i'm getting "incompatible SDP error".<br /><br />In asterisk console,<br /><br />"WARNING[2612][C-00000000]:chan_sip.c:10487 process_sdp:Rejecting secure audio stream without encryption details:audio 18435 RTP/SAVPF 111 103 104 0 8 106 105 13 126<br /> == Using SIP RTP CoS mark 5"<br /><br />but before one month(i.e: April) same configuration works for me <br /><br />Is there any problem with chrome stable or asterisk ?<br /><br />Please help me.<br /><br />Thanks & Regards<br /><br />kudpudeenKudpudeen .Mhttps://www.blogger.com/profile/14712210753696032258noreply@blogger.com