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

[Dmitriy Serov]

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
>