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<p>On 08/09/20 4:16, Joshua C. Colp wrote:<br>
</p>
<blockquote type="cite"
cite="mid:CAM0A2Z0d69m-bdiSXvXc_Lc0RZev8jxiNLrA1gFXjJRWWO6mzA@mail.gmail.com">
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<div dir="ltr">On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <<a
href="mailto:cursor@telecomab.mx" moz-do-not-send="true">cursor@telecomab.mx</a>>
wrote:<br>
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<div class="gmail_quote">
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left:1px solid
rgb(204,204,204);padding-left:1ex"> Some users have
complained that their calls drop after about 30 <br>
seconds. Not all, just some. After looking at the log
files the only <br>
difference I can find from the dropped calls is the
following line:<br>
<br>
[2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c:
Bridge <br>
14410400-5e04-4358-af0c-45fd71f6f5cd: switching from
simple_bridge <br>
technology to native_rtp<br>
<br>
Most calls just do:<br>
<br>
[2020-09-07 18:13:56] VERBOSE[15293][C-00000084]
bridge_channel.c: <br>
Channel PJSIP/1028-0000012a joined 'simple_bridge'
basic-bridge <br>
<626258fc-0649-45c7-b0d3-630a06d2c91b><br>
<br>
Why are some calls using the simple bridge and others
switch to the <br>
native_rtp bridge? Could this be a codec problem? How can
I prevent <br>
the switch?<br>
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<div>It depends on the channels involved as well as the
features in use. To prevent direct media from occurring you
can set the "direct_media" option to "no" on the endpoint.
The native_rtp bridge can still be used, though, to provide
more efficient in-Asterisk forwarding of media. </div>
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<div><br>
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If that doesn't change things you'd need to examine further,
such as looking at the SIP trace for a call (pjsip set logger
on) as 30 seconds is close to the amount of time for a lost ACK
to a 200 OK, which generally indicates a NAT issue.<br
clear="all">
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<p> Direct media is off for all endpoints (both trunks and
phones). There is no NAT on either side, the phones are on the
local network and the trunk provider has a direct link and the pbx
has a dedicated ethernet port for it. We have two trunk providers
and I only see the native rtp bridge used on one of them. I will
do a packet capture on the trunk interface to see if something
else strange happens.</p>
<p> Thank you.<br>
</p>
<pre class="moz-signature" cols="72">--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161</pre>
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