'Re: [asterisk-dev] [Code Review]: Help mitigate reinvite glares in the SIP channel driver'

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List:       asterisk-dev
Subject:    Re: [asterisk-dev] [Code Review]: Help mitigate reinvite glares in the SIP channel driver
From:       "Mark Michelson" <reviewboard () asterisk ! org>
Date:       2012-05-29 20:04:47
Message-ID: 20120529200447.29135.77093 () hotblack ! digium ! com
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> On May 29, 2012, 10:04 a.m., Mark Michelson wrote:
> > /trunk/channels/chan_sip.c, lines 30604-30607
> > <https://reviewboard.asterisk.org/r/1946/diff/2/?file=28284#file28284line30604>
> > 
> > I decided to do another round of testing before committing and noticed that \
> > despite having directmedia=outgoing set, I would see reinvite glares \
> > occasionally. I then noticed this block of code. I had meant to be returning here \
> > instead of continuing with an unlocked sip_pvt and channel. Adding the return \
> > results in one-way audio in my tests, so I need to figure out why that is \
> > happening and how to fix it.

Welp, it looks like this implementation of the patch is not going to work after all.

Consider a situation where phone A places a call through Asterisk 1, which then calls \
through Asterisk 2, which then calls phone B. Direct media is desired. With this \
patch, here is what occurs:

Upon call setup, Asterisk 1 sends a direct media reinvite to phone A saying to send \
media to Asterisk 2. Since the call is outgoing, Asterisk 1 also sends a direct media \
reinvite to Asterisk 2 saying to send media to phone A.

Upon call setup, Asterisk 2 sends a direct media reinvite to phone B saying to send \
media to Asterisk 1. Since the call is outgoing, Asterisk 2 does not send a direct \
media reinvite to Asterisk 1.

Upon receiving the direct media reinvite from Asterisk 1, Asterisk 2 then sends out a \
direct media reinvite to phone B to send media to phone A.

This is the final media setup of the call. Phone B properly sends media to Phone A, \
but Phone A is sending media to Asterisk 2.

I actually see Asterisk 1 send another reinvite each to phone A and to Asterisk 2, \
but the media addresses (c= lines in SDP) are the same as the previous reinvites \
(perhaps this is due to a connected line change?).

The issue here is that Asterisk 1 is never told that the remote media address on the \
outbound call leg has been updated to be phone B. Normally, Asterisk 1 would know of \
this change because of a reinvite being received on the outbound call leg indicating \
the address change. However, reinvites from Asterisk 2 to Asterisk 1 get blocked.

The only way I could get this setup to work properly was if I set nat=comedia so that \
for the brief period where Phone B sends media to Asterisk 1, Asterisk 1 detects a \
media address change and thus reinvites Phone A to send media to Phone B. I had \
strictrtp turned off, so this may not have worked properly with strictrtp enabled.

I'm at a bit of a loss for how to handle this properly. Obviously, Asterisk 2 has to \
be able to send a reinvite to Asterisk 1 so that Asterisk 1 can reinvite phone A \
properly. The obvious answer is to have a setting so that directmedia=outgoing means \
that the outgoing leg reinvites *first* rather than meaning that only the outgoing \
leg reinvites. The thing to work out for this is going to be figuring out how to \
trigger Asterisk 2 in the above scenario to send a reinvite to Asterisk 1. Is it \
enough to complete a single reinvite transaction between Asterisk 1 and 2 before \
Asterisk 2 is allowed to send a reinvite to Asterisk 1? I don't know for sure. I \
think that if there are more than 2 Asterisk servers in between the phones, then \
waiting for a single reinvite transaction might not be enough to effectively reduce \
the chances of glare.

Basically, the long and short of this is that the problem is not as easily solvable \
as what I have hear and will need a more complex solution. I am withdrawing this \
review until I come up with a better solution.

- Mark

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1946/#review6336
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On May 25, 2012, 2:12 p.m., Mark Michelson wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1946/
> -----------------------------------------------------------
> 
> (Updated May 25, 2012, 2:12 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> There are times where multiple Asterisk servers are peered together over SIP. In \
> such situations, it is possible for both Asterisk servers to attempt to send direct \
> media reinvites to each other simultaneously. This results in a glare situation in \
> which each of the Asterisk servers sends a 491 to the other. After a waiting \
> period, the reinvites are re-attempted. This waiting period can potentially be \
> distracting since it can cause the media to take multiple seconds to finalize, \
> especially if more than 2 Asterisk servers are involved. 
> This patch introduces a new SIP peer option called "directmedia_outgoing". If \
> enabled, then when communicating with the peer, Asterisk will only attempt to send \
> reinvites if the call direction is outgoing. The assumption is that the peer \
> Asterisk server will also have this setting enabled. This way, when the two \
> Asterisk servers communicate, they will never attempt to send direct media \
> reinvites to each other. Instead, it will always be the peer that placed the call \
> that will send the direct media reinvite. 
> 
> Diffs
> -----
> 
> /trunk/channels/chan_sip.c 367639 
> /trunk/channels/sip/include/sip.h 367639 
> /trunk/configs/sip.conf.sample 367639 
> 
> Diff: https://reviewboard.asterisk.org/r/1946/diff
> 
> 
> Testing
> -------
> 
> I have tested this by running two Asterisk servers and ensuring that the option was \
> honored and that the media streams were still set up properly. 
> 
> Thanks,
> 
> Mark
> 
> 

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      This is an automatically generated e-mail. To reply, visit:
      <a href="https://reviewboard.asterisk.org/r/1946/">https://reviewboard.asterisk.org/r/1946/</a>
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<blockquote style="margin-left: 1em; border-left: 2px solid #d0d0d0; padding-left: \
10px;">  <p style="margin-top: 0;">On May 29th, 2012, 10:04 a.m., <b>Mark \
Michelson</b> wrote:</p>  <blockquote style="margin-left: 1em; border-left: 2px solid \
#d0d0d0; padding-left: 10px;">  

<table width="100%" border="0" bgcolor="white" style="border: 1px solid #C0C0C0; \
border-collapse: collapse; margin: 2px padding: 2px;">  <thead>
  <tr>
   <th colspan="4" bgcolor="#F0F0F0" style="border-bottom: 1px solid #C0C0C0; \
font-size: 9pt; padding: 4px 8px; text-align: left;">  <a \
href="https://reviewboard.asterisk.org/r/1946/diff/2/?file=28284#file28284line30604" \
style="color: black; font-weight: bold; text-decoration: \
underline;">/trunk/channels/chan_sip.c</a>  <span style="font-weight: normal;">

     (Diff revision 2)

    </span>
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 </thead>

 <tbody style="background-color: #e4d9cb; padding: 4px 8px; text-align: center;">
  <tr>

   <td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static \
int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, \
struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct \
ast_format_cap *cap, int nat_active)</pre></td>

  </tr>
 </tbody>

 <tbody>

  <tr>
    <th bgcolor="#b1ebb0" style="border-right: 1px solid #C0C0C0;" \
align="right"><font size="2"></font></th>  <td bgcolor="#c5ffc4" width="50%"><pre \
style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>  <th \
bgcolor="#b1ebb0" style="border-left: 1px solid #C0C0C0; border-right: 1px solid \
#C0C0C0;" align="right"><font size="2">30604</font></th>  <td bgcolor="#c5ffc4" \
width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">	<span \
class="k">if</span> <span class="p">(</span><span class="n">ast_test_flag</span><span \
class="p">(</span><span class="o">&amp;</span><span class="n">p</span><span \
class="o">-&gt;</span><span class="n">flags</span><span class="p">[</span><span \
class="mi">2</span><span class="p">],</span> <span \
class="n">SIP_PAGE3_DIRECT_MEDIA_OUTGOING</span><span class="p">)</span> <span \
class="o">&amp;&amp;</span> <span class="o">!</span><span class="n">p</span><span \
class="o">-&gt;</span><span class="n">outgoing_call</span><span class="p">)</span> \
<span class="p">{</span></pre></td>  </tr>

  <tr>
    <th bgcolor="#b1ebb0" style="border-right: 1px solid #C0C0C0;" \
align="right"><font size="2"></font></th>  <td bgcolor="#c5ffc4" width="50%"><pre \
style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>  <th \
bgcolor="#b1ebb0" style="border-left: 1px solid #C0C0C0; border-right: 1px solid \
#C0C0C0;" align="right"><font size="2">30605</font></th>  <td bgcolor="#c5ffc4" \
width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">		<span \
class="n">ast_log</span><span class="p">(</span><span \
class="n">LOG_NOTICE</span><span class="p">,</span> <span class="s">&quot;Can&#39;t \
send reinvite because it is an incoming call!</span><span class="se">\n</span><span \
class="s">&quot;</span><span class="p">);</span></pre></td>  </tr>

  <tr>
    <th bgcolor="#b1ebb0" style="border-right: 1px solid #C0C0C0;" \
align="right"><font size="2"></font></th>  <td bgcolor="#c5ffc4" width="50%"><pre \
style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>  <th \
bgcolor="#b1ebb0" style="border-left: 1px solid #C0C0C0; border-right: 1px solid \
#C0C0C0;" align="right"><font size="2">30606</font></th>  <td bgcolor="#c5ffc4" \
width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">		<span \
class="n">sip_pvt_unlock</span><span class="p">(</span><span class="n">p</span><span \
class="p">);</span></pre></td>  </tr>

  <tr>
    <th bgcolor="#b1ebb0" style="border-right: 1px solid #C0C0C0;" \
align="right"><font size="2"></font></th>  <td bgcolor="#c5ffc4" width="50%"><pre \
style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>  <th \
bgcolor="#b1ebb0" style="border-left: 1px solid #C0C0C0; border-right: 1px solid \
#C0C0C0;" align="right"><font size="2">30607</font></th>  <td bgcolor="#c5ffc4" \
width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">		<span \
class="n">ast_channel_unlock</span><span class="p">(</span><span \
class="n">chan</span><span class="p">);</span></pre></td>  </tr>

 </tbody>

</table>

  <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 decided to do another \
round of testing before committing and noticed that despite having \
directmedia=outgoing set, I would see reinvite glares occasionally. I then noticed \
this block of code. I had meant to be returning here instead of continuing with an \
unlocked sip_pvt and channel. Adding the return results in one-way audio in my tests, \
so I need to figure out why that is happening and how to fix it.</pre>  </blockquote>

</blockquote>
<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;">Welp, it \
looks like this implementation of the patch is not going to work after all.