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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3472/">https://reviewboard.asterisk.org/r/3472/</a>
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<p style="margin-top: 0;">On April 29th, 2014, 2:25 p.m. UTC, <b>opticron</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3472/diff/1/?file=57771#file57771line1080" style="color: black; font-weight: bold; text-decoration: underline;">/branches/12/res/res_pjsip_sdp_rtp.c</a>
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(Diff revision 1)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media,</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="cm">/* Ensure the RTP instance is active */</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="n">ast_rtp_instance_activate</span><span class="p">(</span><span class="n">session_media</span><span class="o">-></span><span class="n">rtp</span><span class="p">);</span></pre></td>
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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;">I guess people don't use SRTP+video/text that often...</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">activate isn't really used by res_rtp_asterisk, except for DTLS-SRTP these days</pre>
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<p>- Joshua</p>
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<p>On April 23rd, 2014, 11:06 a.m. UTC, Joshua Colp wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Joshua Colp.</div>
<p style="color: grey;"><i>Updated April 23, 2014, 11:06 a.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-23558">ASTERISK-23558</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;">The code for queueing hold and unhold currently favors unholding over holding. This means that if an SDP is received with host information but with a stream state of sendonly the code will incorrectly stop musiconhold if it has been started.
The attached change fixes this by favoring hold instead of unhold. If an SDP is received with 0.0.0.0 OR sendonly and hold has not been started then it is started. If it has already been started no change occurs. Otherwise if musiconhold has been started it is stopped.</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;">Ran test before fixing and confirmed broken. Ran test after fixing and confirmed fixed.</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_pjsip_sdp_rtp.c <span style="color: grey">(412900)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3472/diff/" style="margin-left: 3em;">View Diff</a></p>
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