[asterisk-users] Some calls drop after 30 seconds

Duncan Turnbull duncan at turnbull.co.nz
Tue Sep 8 05:58:29 CDT 2020

Hi Carlos

On Tue, 8 Sep 2020, 12:36 pm Carlos Chavez, <cursor at telecomab.mx> wrote:

>      Some users have complained that their calls drop after about 30
> seconds.

The rtp timeout is usually about 30 seconds. If rtp is only 1 way then the
calls will drop after 30 secs. This is usually nat/firewall related so a
packet dump helps to confirm. I also find using tcpdump to write a pcap
file that I can feed into wireshark is helpful as wireshark has great sip
decoding options. It will trace the callflow, pull out relevant packets,
replay audio. Its very helpful

Is there anything different about these users and their setup? Or who they
are calling?

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