[asterisk-dev] One sip stack to rule them all....

[Bryant Zimmerman]

I would agree with this. We have tried to deploy pjsip several times over the last year with limited success. 
 We have had nothing but issues with database real-time deployments. Tables not working from one 13.x release to another. 
 Table builders sorcery failing out.  
  
 Issues when there are multiple transports on varying networks were udp is not routed correctly through the asterisk servers. No matter the settings. 
  
 Connectivity issues with varying success by carrier. 
  
 Unexplained audio quality issues that don't occur on the same spec running chan_sip
  
 We want to move to pjsip but the functionality and stability have only proven out for limited applications.
  
  
 Bryant
  

----------------------------------------
 From: "Daniel Journo" <dan at keshercommunications.com>
Sent: Sunday, October 8, 2017 3:12 PM
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Subject: Re: [asterisk-dev] One sip stack to rule them all....   

> What is _also_ needed, however, is more use of PJSIP and reports of  specific problems, and specific deficits of PJSIP so that the fear can be eased before, at some point many years from now, chan_sip just doesn't work any more.

There are a number of specific issues on issue tracker which still need addressing before more people will take it on properly. Some issues probably require a semi-major rethink and probably won't be dealt with for months.
Making chan_sip depreciated would leave Asterisk with no production grade sip stack that is officially being maintained. 


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