On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <[email protected]> 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. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
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