[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
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