<div dir="ltr"><div class="gmail_extra"><div class="gmail_quote">2015-05-21 18:43 GMT+02:00 Joshua Colp <span dir="ltr"><<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Ludovic Gasc wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span class="">
2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <<a href="mailto:jd.girard@sysnux.pf" target="_blank">jd.girard@sysnux.pf</a><br></span>
<mailto:<a href="mailto:jd.girard@sysnux.pf" target="_blank">jd.girard@sysnux.pf</a>>>:<span class=""><br>
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Le 21/05/2015 00:16, Joshua Colp a écrit :<br>
> If CCSS is needed then the only option is to use chan_sip. The<br>
> chan_pjsip module does not implement CCSS in any way.<br>
<br>
Is CCSS support planned for PJSIP? chan_sip is in "extended" state in<br>
asterisk-13, so chan_pjsip should be preferred for new installations, ri<br>
ght?<br>
<br>
<br>
If you really want CCSS support and to be fancy with PJSIP, you can<br>
easily implement a similar feature with AMI events, I already did that a<br>
long time ago before the integration of CCSS in Asterisk.<br>
I think it's possible to implement that only with dialplan and call files.<br>
<br>
In my mind, chan_sip will be dropped after asterisk 13, is it true ?<br>
</span></blockquote>
<br>
It won't be dropped. It still has features which are not available in PJSIP, and people still use it. The extended status refers to the support level. Per the support states wiki page[1]:<br>
<br>
This module is supported by the Asterisk community, and may or may not have an active developer. Some extended modules have active community developers; others do not. Issues reported against these modules may have a low level of support.<br></blockquote><div><br></div><div>Joshua, come on, you know as me that you have few people around the world to have the skills and the time to maintain a C module for Asterisk.</div><div>For a critical feature like SIP in Asterisk, at least to me, it means that for a serious production with Asterisk 13, I won't use chan_sip but I'll prefer chan_pjsip.</div><div>Personally, I don't care if it's pjsip or sip, I only want a telephony stack that won't piss on my shoes under the fire of a big production.</div><div><br></div><div>However, I didn't know that some features are missing in chan_pjsip compare to chan_sip. A list exists somewhere ?</div><div><br></div><div>Moreover, by curiosity, somebody has already benchmarked chan_sip vs chan_pjsip ? Somebody has a noticed an efficiency issue with pjsip ?</div><div><br></div><div>Regards.</div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
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[1] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States</a><span class="im"><br>
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-- <br>
Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>
Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
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