[asterisk-dev] [Code Review] Add RTP keep-alives back to Asterisk 1.8 after they were accidentally removed when moving to the RTP Engine API.

[Terry Wilson]

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https://reviewboard.asterisk.org/r/1226/
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Review request for Asterisk Developers.


Summary
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The move to RTP engine left out RTP keep-alives. This is a port of RTP keep-alives from 1.6.2 to the RTP Engine API.


This addresses bug 18697.
    https://issues.asterisk.org/view.php?id=18697


Diffs
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  /branches/1.8/channels/chan_sip.c 320161 
  /branches/1.8/include/asterisk/rtp_engine.h 320161 
  /branches/1.8/main/rtp_engine.c 320161 
  /branches/1.8/res/res_rtp_asterisk.c 320161 

Diff: https://reviewboard.asterisk.org/r/1226/diff


Testing
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sip.conf
[global]
rtpkeepalive=1 ; second

extensions.conf
[default]
exten => test2,1,Answer
exten => test2,n,Playback(vm-goodbye)
exten => test2,n,Wait(10)

*CLI> rtp set debug on

sipp -sn uac 127.0.0.1 -s test2 -d 10000 -m 1 -rtp_echo

Verified the CLI shows once per second: Sent Comfort Noise RTP packet to 127.0.1.1:6000 (type 02, seq 062924, ts 008160, len 000001)


Thanks,

Terry

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