On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote: > Yes, this means the remote end was not sending any audio stream. > But it shouldn't. > According to [1] before remote end send OK or ACK there is one way SDP, > no any audio stream. > PJSIP channel (option rtp_timeout) does not take this one. > > Isn't it a mistake? What could be workarounds?
Looking at the code we don't take that scenario into account it seems. Please file an issue[1] and we'll see if we can do something about it. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
