[asterisk-dev] chan_sip deprecation

Jon Bonilla (Manwe) manwe at aholab.ehu.es
Tue Nov 22 02:13:52 CST 2022


El Tue, 22 Nov 2022 08:00:48 +0000
Henning Westerholt <hw at gilawa.com> escribió:

> Hello,
> 
> I am really wondering why people are trying to keep chan_sip alive. No
> offence to the past developers, but pjsip is a much better SIP stack
> regarding standard compliance and stability compared to the old one. Also,
> regarding performance chan_pjsip is better. From an outside view, the
> asterisk project gave plenty of time to migrate.
> 

Not defending here keeping chan_sip, it will be removed and chan_pjsip will
need to be adopted. But just from my point of view as asterisk based solution
developer:

We know chan_sip. We know what works and what fails. And we know the
workarounds for what fails. Having hundreds of asterisk servers working 24/7
during  the last 7 years I had 0 crashes using chan_sip (don't saying here that
chan_pjsip would have been different).

No need of new features here as I use kamailio for some stuff like path,
paralel forking, websocket handling and all kind of stuff. Just want chan_* to
send/receive calls fast. 

chan_pjsip probably hasn't routed yet 1% of the calls chan_sip routed in all
history. 

So, In my case it's more confortable to keep chan_sip as I don't need anything
else and I have 0 issues with it. Maybe others are in the same position.


But IMHO chan_sip must be removed. I understand what the development is and
it's a PITA to keep old code and even keeping it unmantained is a pain. When
the time comes I'll move to chan_pjsip and I won't complain. Grateful to the
asterisk devs that provide such a great solution.

cheers,

Jon



-- 
PekePBX, the multitenant PBX solution
https://pekepbx.com



More information about the asterisk-dev mailing list