Ive configured TURN in rtp.conf in Asterisk 11.5.The credentials are correct because I can get Chrome to get relay candidates and attach them to the SDP, but Asterisk doesnt want to play ball.Theres little documentation — at least from what I can t..
I was wondering if anyone knows if Asterisk 12 will be supporting the OPUScodec, which is part of the WebRTC standard as the default codec…
Im working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org/jira/browse/ASTERISK-21696Currently, Im systematically going through each Makefile in every direct..
After struggling with one way audio issues as a result of STUN binding errors on both the Asterisk side and the Chrome side, weve decided to just simply go with a TURN relay for RTP packets until the issues are resolved.I configured rtp.conf so t..