On Tue, 2019-01-29 at 10:54 +0100, Bernhard Schmidt wrote: > Hi James, > > thanks. Have you raised this issue with upstream somehow? I know > chan_sip is deprecated, but I doubt a bug this severe would be > undetected for that long. > > I'll try to whip together a test for this (my test installation is > using chan_pjsip and IPv6).
Actually, you can close this; it turns out to be a firewall issue: the asterisk 16 update apparently changed the rtp.conf file, moving the RTP listening port range outside of the matching range on the firewall hence an intermittent audio loss problem with external connections. What the patch does is mostly open the conntrack port on the firewall allowing the inbound RTP stream. The correct fix is, of course, to put the rules back to matching each other. James
