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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1663/">https://reviewboard.asterisk.org/r/1663/</a>
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<p style="margin-top: 0;">On January 13th, 2012, 12:25 p.m., <b>Matt Jordan</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1663/diff/2/?file=23157#file23157line543" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/res/res_rtp_asterisk.c</a>
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(Diff revision 2)
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<td colspan="2"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
<td colspan="2"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int rtp_learning_rtp_seq_update(struct ast_rtp *rtp, int seq/*struct learning_rtp_status *seq_status*/)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="p">}</span> <span class="k">else</span> <span class="k">if</span> <span class="p">(</span><span class="n">udelta</span> <span class="o">==</span> <span class="mi">0</span><span class="p">)</span> <span class="p">{</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="p">}</span> <span class="k">else</span> <span class="k">if</span> <span class="p">(</span><span class="n">udelta</span> <span class="o"><</span> <span class="n">LEARNING_MAX_DROPOUT</span><span class="p">)</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;">As you have it now, none of the udelta checking will ever get called. If rtp->learning_probation is non-zero, we don't check any of it - and rtp->learning_probation should never be zero in this method as, as soon as it is 0, we set the learning mode to STRICT_RTP_CLOSED. I'm not sure if this is what you expected or not, but it looks like the code that is executed when we receive a non-expected sequence number is not sufficient to account for all of the things that the pjmedia code was taking into account.</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;">You are correct on the probation never being meant to be zero. If anything in there is expecting a probation of 0, I'll go ahead and cull it since it was just a remnant from the pjmedia equivalent of this function.</pre>
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<p style="margin-top: 0;">On January 13th, 2012, 12:25 p.m., <b>Matt Jordan</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1663/diff/2/?file=23157#file23157line552" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/res/res_rtp_asterisk.c</a>
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(Diff revision 2)
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<td colspan="2"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int rtp_learning_rtp_seq_update(struct ast_rtp *rtp, int seq/*struct learning_rtp_status *seq_status*/)</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;">Blob</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;">Beginning to wish I'd taken the time to set up my vimrc on that machine a little better.</pre>
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<p style="margin-top: 0;">On January 13th, 2012, 12:25 p.m., <b>Matt Jordan</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1663/diff/2/?file=23157#file23157line2361" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/res/res_rtp_asterisk.c</a>
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(Diff revision 2)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="n">ast_verb</span><span class="p">(</span><span class="mi">3</span><span class="p">,</span> <span class="s">"seqno set</span><span class="se">\n</span><span class="s">"</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;">This isn't a terribly useful verbose message. Since that's something users will see more often, it should be either something specific that is useful to them, or a debugging message (ideally it should include the sequence number that was set, as well as the RTP session pointer, which you referenced previously)</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;">That's just a remnant of something. Thanks for catching it.</pre>
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<p>- jrose</p>
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<p>On January 13th, 2012, 11:16 a.m., jrose wrote:</p>
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<div>Review request for Asterisk Developers, Kevin Fleming, Matt Jordan, and jcolp.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated Jan. 13, 2012, 11:16 a.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;">The purpose of this patch is to resolve some problems that can occur with peers using a mix of directmedia=yes and no occuring between multiple asterisk servers when strictrtp is enabled in rtp.conf. When attempting to set the strictrtppeer during a reinvite, res_rtp_asterisk will ignore the requested sockaddr and instead set learning mode. Prior to this patch, learning mode would simply set the sockaddr owning the next related RTP packet received and then exit learning mode. Now, learning mode will track how many times a particular socket address sent rtp packets without being interrupted by another socket address until it reaches the learning mode hit threshold value (set in rtp.conf) and then resume closed mode.</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 tested this against two scenarios with the following configurations... always with strictrtp=yes
s1
Phone 1 <-- DirectMedia=yes --> Asterisk 1 <-- DirectMedia=Yes --> Asterisk 2 <-- DirectMedia=No --> Phone 3
and
s2
Phone 1 <-- DirectMedia=yes --> Asterisk 1 <-- DirectMedia=Yes --> Phone 2
and then doing a blind transfer with:
Phone 2 <-- DirectMedia=yes --> Asterisk 1 <-- DirectMedia=Yes --> Asterisk 2 <-- DirectMedia=No --> Phone 3
Prior to this test, in s1 there would be one way audio from phone 3 to phone 1 because RTP coming from phone 1 would be rejected by Asterisk 2.
In s2, audio would work normally from phone 1 to phone 2, but after the transfer the result would be the same with one way audio from phone 3 to phone 1 due to rejected
packets from phone 1.
After this patch is applied, both of these scenarios worked as expected as long as I used a threshold above 3. Phones 1 and 3 were polycom SP430s while Phone 2 was a Grandstream GXP-2020.</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/1.8/res/res_rtp_asterisk.c <span style="color: grey">(350549)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1663/diff/" style="margin-left: 3em;">View Diff</a></p>
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