]> git.ipfire.org Git - thirdparty/asterisk.git/commit
chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active
authorMichael Neuhauser <mike@firmix.at>
Fri, 6 Mar 2020 16:50:00 +0000 (17:50 +0100)
committerGeorge Joseph <gjoseph@digium.com>
Fri, 20 Mar 2020 15:17:33 +0000 (10:17 -0500)
commit580e260ff841e966b7eb2f1609186b4db5bfe741
treee3c41a0d51ca31452aedd95c78848fd92a1aefea
parentc7c52b8e7ab2c4e1e27bd06b7c35a9ba96711bf2
chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active

Do not hang up a PJSIP channel on RTP timeout if that channel is in
a direct-media bridge. Also reset the time of the last received RTP packet when
direct-media ends (wait full rtp_timeout period before checking first time after
audio came back to Asterisk).

ASTERISK-28774
Reported-by: Michael Neuhauser
Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1
channels/chan_pjsip.c
res/res_pjsip_sdp_rtp.c