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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1226/">https://reviewboard.asterisk.org/r/1226/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By Terry Wilson.</div>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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)</pre>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/view.php?id=18697">18697</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/1.8/channels/chan_sip.c <span style="color: grey">(320161)</span></li>
<li>/branches/1.8/include/asterisk/rtp_engine.h <span style="color: grey">(320161)</span></li>
<li>/branches/1.8/main/rtp_engine.c <span style="color: grey">(320161)</span></li>
<li>/branches/1.8/res/res_rtp_asterisk.c <span style="color: grey">(320161)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1226/diff/" style="margin-left: 3em;">View Diff</a></p>
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