[asterisk-dev] Video calling

Matt Fredrickson creslin at digium.com
Mon Nov 18 15:43:41 CST 2019


On Fri, Nov 15, 2019 at 7:38 PM Troy Bowman <troy at lump.net> wrote:
>
> On Fri, Nov 15, 2019 at 3:56 PM John Kiniston <johnkiniston at gmail.com> wrote:
>>
>> I do not recommend using chan_sip, chan_sip is no longer receiving development.
>> chan_pjsip is where the development focus is at.
>
>
> Sure, chan_pjsip is where the feature development focus is, but is it truly stable enough for production now?  It seems I still see a lot of bug fixes for seemingly constant problems, while chan_sip's code is so mature that it just works hands-off.  I'm still afraid of using chan_pjsip in production just like I am still afraid of Linux's btrfs in production, and btrfs has been in development for over a decade.

Sangoma/Digium has been using chan_pjsip exclusively in Switchvox for
a few years now.

Also, chan_sip doesn't really receive bug fixes anymore, so there's
that problem as well.  With regards to the poster's original question,
I'd definitely suggest using chan_pjsip in a new video based
environment as that is the channel driver that has been tested and
used with all the new video work in Asterisk.

-- 
Matthew Fredrickson
Digium - A Sangoma Company | Director of Open Source Software Development
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA



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