El Tue, 22 Nov 2022 08:00:48 +0000 Henning Westerholt <h...@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 -- _____________________________________________________________________ -- 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