[asterisk-users] Some calls drop after 30 seconds
Joshua C. Colp
jcolp at sangoma.com
Tue Sep 8 04:16:37 CDT 2020
On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <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[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[C-00000084] bridge_channel.c:
> Channel PJSIP/1028-0000012a joined 'simple_bridge' basic-bridge
> 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.
Joshua C. Colp
Asterisk Technical Lead
Check us out at www.sangoma.com and www.asterisk.org
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