[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
jcolp at digium.com
Tue Jan 3 11:58:21 CST 2017
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  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 and we'll see if we can do something about it.
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
More information about the asterisk-users