[Asterisk-code-review] chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is ac... (asterisk[13])

[Michael Neuhauser]

Hello Friendly Automation, 

I'd like you to reexamine a change. Please visit

    https://gerrit.asterisk.org/c/asterisk/+/13976

to look at the new patch set (#2).

Change subject: chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active
......................................................................

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
---
M channels/chan_pjsip.c
M res/res_pjsip_sdp_rtp.c
2 files changed, 41 insertions(+), 10 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/76/13976/2
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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1
Gerrit-Change-Number: 13976
Gerrit-PatchSet: 2
Gerrit-Owner: Michael Neuhauser <mike at firmix.at>
Gerrit-Reviewer: Friendly Automation
Gerrit-MessageType: newpatchset
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