<span style="font-family: Arial, Helvetica, Sans-Serif; font-size: 12px"><div>I agree the multi-domain environment is a nice idea, but too many endpoints don't properly support.<br />
We to use a prefix in the SIP username for multi-domain environments.</div>
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Thanks<br />
Bryant</div>
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<div><span style="font-family: tahoma,arial,sans-serif; font-size: 10pt;"><b>From</b>: "Ludovic Gasc" <gmludo@gmail.com><br />
<b>Sent</b>: Sunday, July 17, 2016 5:20 PM<br />
<b>To</b>: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br />
<b>Subject</b>: Re: [asterisk-users] PJSIP - State of the art</span>
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<div class="gmail_quote">2016-07-17 14:30 GMT+02:00 Annus Fictus <span dir="ltr"><<a href="mailto:annusfictus@gmail.com" target="_blank">annusfictus@gmail.com</a>></span>:
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<p>The main idea of the new channel was working on a multi-domain environment</p>
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<div>For now, to my experience, it's more future-proof compliant to use a prefix in the SIP username than multi-domain environment.</div>
<div>Even if the multi-domain support was perfect in Asterisk, we tested some crappy SIP endpoints where in fact, even if you configure a domain name everywhere in the configuration, you have only IPs in SIP packets.</div>
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<div>We have that on production for our cloud plateform, it works pretty well and also simplify whitelabel handling.</div>
<div>Moreover, if you have a good provisioning support, it will be invisible for your users.</div>
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<div>When I see the time needed to really use on production the SNI feature in SSL, and you have only 5 majors HTTP endpoints (aka Web browsers).</div>
<div>In the SIP world, I'm not sure you can use multi domain except if you can force the SIP endpoints used by your clients.</div>
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<p>, have more then one device registered with same credentials and have more stability.</p>
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<div>Since 13.9.1, we have a better experience of pjsip.</div>
<div>Nevertheless, not yet massively used on production for now, we planned to migrate endpoint by endpoint to minimize the risk. </div>
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<p>Be Better still with Asterisk 1.11.X?</p>
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<div>Maybe you could use Asterisk 13 with chan_sip to start, it works pretty well and already think to support chan_pjsip in the same time.</div>
<div>The benefit to think about that if one day you need to use an alternative channel like chan_iax2, it should be easier to implement for you.</div>
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<p>Regards</p>
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