Joshua, issue has been filed. Thank you!

https://issues.asterisk.org/jira/browse/ASTERISK-26689

03.01.2017 20:58, Joshua Colp пишет:
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



--
_____________________________________________________________________
-- 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

Reply via email to