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<p>Hi</p>
<p><br>
</p>
<p>I have spent over a year migrating from chan_sip (1.8) to chan_pjsip (13) and it has been stressful. </p>
<p><br>
</p>
<p>However, there is light at the end of the tunnel. When first migrating Asterisk would crash around 20 times a day or more. However, by investing time and money into resolving the segfaults, database issues and task managers I feel that the new stack is stable
with the odd bug still remaining. The most common crash I get from the stack at the moment is due to TLS connections, which the PJSIP team are currently working on and I am assured there will be a patch in the coming days.</p>
<p><br>
</p>
<p>From experience, I can say that chan_pjsip is more scalable and efficiently uses server resources compared to chan_sip. It is the way forward!</p>
<p><br>
</p>
<p>I would welcome a working group to manage the migration from chan_sip to chan_pjsip as there are still features in chan_sip that have not been implemented in chan_pjsip. I would also welcome additional features such as '<span>Device Feature Key Synchronization'
(as-feature-event).</span></p>
<p><span><br>
</span></p>
<p><span>At present, there are a few undocumented features, such as the sorcery configuration:</span></p>
<p><span><br>
</span></p>
<blockquote style="margin: 0 0 0 40px; border: none; padding: 0px;">
<p><span><span>endpoint=realtime,ps_endpoints,<b>allow_unqualified_fetch=error</b></span></span></p>
<p><span><span><b><br>
</b></span></span></p>
</blockquote>
The above stops a full database query that loads every single endpoint at startup, which can cause overload on systems with a number of endpoints. Therefore documentation covering the whole sip stack and features would help people migrate easier.<br>
<p><br>
</p>
<p>Finally, I would like to thank everyone who has been working on ironing out the chan_pjsip bugs.</p>
<p><br>
</p>
<p>Ross</p>
<br>
<br>
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<div id="x_divRplyFwdMsg" dir="ltr"><font face="Calibri, sans-serif" color="#000000" style="font-size:11pt"><b>From:</b> asterisk-dev-bounces@lists.digium.com <asterisk-dev-bounces@lists.digium.com> on behalf of Olle E. Johansson <oej@edvina.net><br>
<b>Sent:</b> 05 October 2016 10:42<br>
<b>To:</b> Asterisk Developers Mailing List<br>
<b>Cc:</b> Olle E Johansson<br>
<b>Subject:</b> Re: [asterisk-dev] Viva Chan_Sip, may it rest in peace</font>
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<div class="PlainText">Hi!<br>
<br>
>From my perspective I know that maintaining a SIP stack requires *A LOT* of effort, so I understand that a project can’t maintain two of them.<br>
<br>
I suggest that a working group is created for the transition and that the first task is to compare the functionality.
<br>
Last time I checked the functionality *I need* (but maybe not everyone else) was non-existing in PJSIP so I could not use it.<br>
It may have changed since then.<br>
<br>
I think the goal has to be to gradually phase out the ugly code in chan_sip and celebrate the day it’s gone, but<br>
make sure we don’t leave functionality (and users) behind and have good guidelines for the transition.<br>
<br>
I still think we should totally rewrite how chan_pjsip is configured. That concept is very far away from other SIP implementations.<br>
But that’s my personal opinion from a small cold corner of the world, using Asterisk in non-PBX ways as large scale media
<br>
and feature servers.<br>
<br>
Executive summary: Create a working group that maintains the feature gap, makes sure it’s going away and also makes sure<br>
that we have enough material that explains the gold that hides in chan_pjsip!<br>
<br>
/O<br>
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