[asterisk-users] Some calls drop after 30 seconds

Carlos Chavez cursor at telecomab.mx
Tue Sep 8 10:45:26 CDT 2020


On 08/09/20 4:16, Joshua C. Colp wrote:

> On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <cursor at telecomab.mx 
> <mailto:cursor at telecomab.mx>> wrote:
>
>          Some users have complained that their calls drop after about 30
>     seconds.  Not all, just some.  After looking at the log files the
>     only
>     difference I can find from the dropped calls is the following line:
>
>     [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge
>     14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge
>     technology to native_rtp
>
>          Most calls just do:
>
>     [2020-09-07 18:13:56] VERBOSE[15293][C-00000084] bridge_channel.c:
>     Channel PJSIP/1028-0000012a joined 'simple_bridge' basic-bridge
>     <626258fc-0649-45c7-b0d3-630a06d2c91b>
>
>          Why are some calls using the simple bridge and others switch
>     to the
>     native_rtp bridge?  Could this be a codec problem?  How can I prevent
>     the switch?
>
>
> 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.
>
> 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.
>
>
     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.

     Thank you.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200908/668729dc/attachment.html>


More information about the asterisk-users mailing list