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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1542/">https://reviewboard.asterisk.org/r/1542/</a>
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<p style="margin-top: 0;">On October 26th, 2011, 6:48 p.m., <b>Alec Davis</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1542/diff/1/?file=21405#file21405line235" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/res/res_rtp_multicast.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 multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="cm">/* Increment sequence number and reset to 0 if it would overflow allocated 8 bits in RTP packet */</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="n">multicast</span><span class="o">-></span><span class="n">seqno</span> <span class="o">=</span> <span class="p">(</span><span class="n">multicast</span><span class="o">-></span><span class="n">seqno</span> <span class="o">+</span> <span class="mi">1</span><span class="p">)</span> <span class="o">%</span> <span class="mi">65535</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;">also the comment seems wrong. shouldn't it be 16 bits in RTP packet.
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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;">Yeah, that's embarassing. 16 is right.</pre>
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<p>- jrose</p>
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<p>On October 26th, 2011, 3:35 p.m., jrose wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated Oct. 26, 2011, 3:35 p.m.</i></p>
<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;">Basically what was happening was the sequence number would go above 65535, and rather than rolling back to zero, the extra bits would overflow into the codec bits of the RTP packet. Because of this, if the sequence number goes above 65535, it'll make the codec value change and this can make calls fail.
This patch simply checks the value of the sequence number each time it is incremented and sets it back to 0 if it's above 65535. Shouldn't cause any problems.</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;">I checked the unpatched version with wireshark to make sure the codec shift was in fact happening and it was. I checked it again after patching and made sure the problem went away and it did. I also checked to see whether the values stayed as expected if I forcibly changed the codec value to something other than 0, and in all cases it behaved as expected.</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-18291">ASTERISK-18291</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/res/res_rtp_multicast.c <span style="color: grey">(341429)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1542/diff/" style="margin-left: 3em;">View Diff</a></p>
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