[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

Joshua Colp 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 [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

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