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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4216/">https://reviewboard.asterisk.org/r/4216/</a>
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<div>Review request for Asterisk Developers and Joshua Colp.</div>
<div>By Kevin Harwell.</div>
<p style="color: grey;"><i>Updated Dec. 4, 2014, 3:31 p.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</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;">Changed to rename things only be internal to the rtp_engine source. Any externally needed api calls were added instead of changed.</pre>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24563">ASTERISK-24563</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<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;">When endpoints with direct_media enabled, behind a firewall (Asterisk on a separate network) and were bridged sometimes Asterisk would send the ip address of the firewall in the sdp to one of the phones in the reinvite resulting in one way audio. When sending the reinvite Asterisk will retrieve the media address from the associated rtp instance, but if frames were being read this can be overwritten with another address (in this case the firewall's). This patch ensures that Asterisk uses the original device address when using direct media.</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;">Used a test bed of 3 phones on a private network behind a firewall with Asterisk on another network. Enabled direct media on the endpoints and then had phone A call phone B. Noted in the logged SIP reinvites that the correct address was now being used and also made sure audio flowed in both directions.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> (updated)</h1>
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<li>branches/12/res/res_rtp_asterisk.c <span style="color: grey">(428862)</span></li>
<li>branches/12/main/rtp_engine.c <span style="color: grey">(428862)</span></li>
<li>branches/12/include/asterisk/rtp_engine.h <span style="color: grey">(428862)</span></li>
<li>branches/12/channels/chan_sip.c <span style="color: grey">(428862)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4216/diff/" style="margin-left: 3em;">View Diff</a></p>
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