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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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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Where's the change for PJSIP? Or was one not necessary?</pre>
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<a href="https://reviewboard.asterisk.org/r/4216/diff/4/?file=69538#file69538line870" style="color: black; font-weight: bold; text-decoration: underline;">branches/12/include/asterisk/rtp_engine.h</a>
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(Diff revision 4)
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> * \brief Set the requested target address of the remote endpoint</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> *</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> * \param instance The RTP instance to change the address on</span></pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I think it is worth explaining here what the difference is between the 'incoming source address' and the 'requested target address'. Even if it is just a sentence or two.
You have that in the 'get' accessors, but it is worth spelling it out here as well.</pre>
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<p>- Matt Jordan</p>
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<p>On December 4th, 2014, 3:31 p.m. CST, Kevin Harwell wrote:</p>
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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>
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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> </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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