28 Feb
2019
28 Feb
'19
9:43 a.m.
El jue., 28 de feb. de 2019 a la(s) 09:21, Daniel Tryba (d.tryba(a)pocos.nl)
escribió:
You wrote something else in your original message:
exposing asterisk RTP 10000-30000 makes it function.
that's the problem here! yes!
I read that to be an indication that rtp(proxy|engine)
isn't rewriting.
in fact, the first try was with rtpproxy, but due the DEBUG log are very
poor
we tried with rtpengine, and now i noted that rtpengine only output in log
that:
```
Feb 28 14:39:44 ip-10-10-1-1 rtpengine[28721]: DEBUG: timer run time =
0.000005 sec
Feb 28 14:39:44 ip-10-10-1-1 rtpengine[28803]: [1551364784.000249] DEBUG:
timer run time = 0.000035 sec
Feb 28 14:39:45 ip-10-10-1-1 rtpengine[28864]: DEBUG: timer run time =
0.000010 sec
Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28864]: DEBUG: timer run time =
0.000007 sec
Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28732]: DEBUG: timer run time =
0.000008 sec
Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28832]: DEBUG: timer run time =
0.000005 sec
```
BUT: rtpproxy seems are working cos if not there's no SIP sesion
stablished!
but as i siad, no sound if i not exposed the asterisk ports!
NOTE: audio of the call are udp! so that means are not possible? i guess
must be possible
due the kamailio can manage an route using the rtp[proxy|engine] right?
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