[asterisk-dev] One sip stack to rule them all....
Troy Bowman
troy at lump.net
Sun Oct 8 11:55:49 CDT 2017
I sincerely hope they don't deprecate it. The pjsip code might seem fine
in development and test environments, but I am still afraid of using it in
production. I see too many issues with it regularly on this list. I can't
gamble stability versus my job security.
>From my perspective, chan_sip doesn't get bugfixes because it doesn't seem
to need them. It just works. I have had zero issues with it for several
years.
On Sun, Oct 8, 2017 at 8:55 AM, James Finstrom <jfinstrom at gmail.com> wrote:
> One does not simply depricate a sip stack.
>
> Ok so at devcon there was a discussion of depricating chan_sip. This may
> sound a lot worse than it actually is. Chan_sip has been essentially
> untouched in 4ish years. It does not receive bug fixes. It is just sort of
> a barge floating in the ocean.
>
> So one of the things that is needed to finally put Chan sip to bed is
> feature parody. Someone brought up CCSS.
>
> What features do you feel you would lose going from chan_sip to pjsip.
>
> Are there any bugs in pjsip that keep you from migrating?
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20171008/3ca8ee06/attachment.html>
More information about the asterisk-dev
mailing list