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

Dmitriy Serov serov.d.p at gmail.com
Tue Jan 3 12:24:06 CST 2017

Joshua, issue has been filed. Thank you!


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

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