Recently, I've had to be on a lot of long conference calls and have noticed that after about 18-22 minutes, the audio on the call drops out. There are no errors logged from Asterisk, Kamailio, or RTPengine. The signalling portion of the call is unaffected and continues as normal.
My setup is as follows asterisk-14.3.0-7.gita047b25.fc25.x86_64 (14 branch) kamailio-5.0.0-5.git2608015.fc25.x86_64 (5.0 branch) kernel-4.10.8-200.fc25.x86_64 (stable Fedora kernel) rtpengine-5.1.1.1-9.git27af349.fc25.x86_64 (master branch) Yealink or Zoiper softphones Calls are originated as follows using TLS & SRTP Yealink/Zoiper -> Kamailio/RTPengine+kernel -> Asterisk/DAHDI/PSTN Related IPtables output below (generated by firewalld) ~]# iptables -S |grep INPUT -P INPUT ACCEPT -N INPUT_ZONES -N INPUT_ZONES_SOURCE -N INPUT_direct -A INPUT -m conntrack --ctstate RELATED,ESTABLISHED -j ACCEPT -A INPUT -i lo -j ACCEPT -A INPUT -j INPUT_direct -A INPUT -j INPUT_ZONES_SOURCE -A INPUT -j INPUT_ZONES -A INPUT -m conntrack --ctstate INVALID -j DROP -A INPUT -j REJECT --reject-with icmp-host-prohibited -A INPUT_direct -p udp -m udp --dport 30000:40000 -j RTPENGINE --id 0 -A INPUT_ZONES_SOURCE -m set --match-set mss4 src -g IN_mss -A INPUT_ZONES -g IN_public -A IN_public -j IN_public_allow -A IN_public_allow -p udp -m udp --dport 30000:40000 -m conntrack \ --ctstate NEW -j ACCEPT I saw https://github.com/sipwise/rtpengine/issues/102 and https://github.com/sipwise/rtpengine/issues/108, but do not seem to have any issues (that I know of) with the SRTP, though the latter ticket references "call that would last longer than 20 minutes". Can you share suggestions for looking into the cause of the audio drops, especially as nothing appears in the logs? Thanks. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6
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