[asterisk-users] asterisk pjsip webrtc rtp to private IP

[marek]

https://issues.asterisk.org/jira/browse/ASTERISK-28656


Dne 12/12/2019 v 16:49 Joshua C. Colp napsal(a):
> On Thu, Dec 12, 2019 at 11:40 AM marek <cervajs64 at gmail.com 
> <mailto:cervajs64 at gmail.com>> wrote:
>
>     examples of "interesting" information like ICE result and howto
>     make "minimal" configuration of pjproject.conf
>
>     i.e.
>
>     for  debugging app_queue.so
>
>     core set debug 5 app_queue.so
>
>     for debugging RTP
>
>     core set debug 10 rtp_engine
>     core set debug 10 res_rtp_asterisk
>
>     rtp set debug on
>
>     logger.conf
>
>     rtp => debug,verbose(5)
>
>
>     so i mean
>
>     in
>     https://github.com/asterisk/asterisk/blob/master/configs/samples/pjproject.conf.sample
>
>     by few examples try to explain  what usefull info i can get
>
>
>     set
>
>     [startup]
>     log_level=6
>     type=startup
>
>     and dig what's usefull is not very productive
>
>     btw we are using tools like sipcapture.org
>     <http://sipcapture.org>,voipmonitor.org <http://voipmonitor.org>,
>     callstats.io <http://callstats.io>, elasticsearch+filebeat, ...
>     but without informations whats happening inside asterisk is harder
>     to solve problems
>
>
> Please file an issue[1] with these specific examples and any others so 
> it's not lost.
>
> [1] https://issues.asterisk.org/jira
>
> -- 
> Joshua C. Colp
> Senior Software Developer
> Sangoma Technologies
> Check us out at www.sangoma.com <http://www.sangoma.com> and 
> www.asterisk.org <http://www.asterisk.org>
>
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