Hi Carlos On Tue, 8 Sep 2020, 12:36 pm Carlos Chavez, <[email protected]> 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 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
