[asterisk-users] PJSIP - State of the art

[Bryant Zimmerman]

I agree the multi-domain environment is a nice idea, but too many endpoints 
don't properly support.
We to use a prefix in the SIP username for multi-domain environments.

Thanks
Bryant
  

----------------------------------------
 From: "Ludovic Gasc" <gmludo at gmail.com>
Sent: Sunday, July 17, 2016 5:20 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] PJSIP - State of the art   
   2016-07-17 14:30 GMT+02:00 Annus Fictus <annusfictus at gmail.com>:    

The main idea of the new channel was working on a multi-domain environment 

  For now, to my experience, it's more future-proof compliant to use a 
prefix in the SIP username than multi-domain environment.
 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.
  
  We have that on production for our cloud plateform, it works pretty well 
and also simplify whitelabel handling.
 Moreover, if you have a good provisioning support, it will be invisible 
for your users.
   

  
 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).
 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.

, have more then one device registered with same credentials and have more 
stability. 
  Since 13.9.1, we have a better experience of pjsip.
 Nevertheless, not yet massively used on production for now, we planned to 
migrate endpoint by endpoint to minimize the risk. 

   

Be Better still with Asterisk 1.11.X? 
  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.
 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.

   

Regards  

   

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users 


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160718/2b3b31f8/attachment.html>