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
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