I have read through all the articles on the two flavours of SIP, namely PJSIP and Chan_SIP. I have read all the stories a few years back about how PJSIP was not stable yet etc, how Chan_SIP is being phased out…
Here is my question, because of a huge crash oh my PBX server, I am rebuilding my FPBX server. I am using FPBX 14 and Asterisk 13. (Unless there is a huge reason why I should use Asterisk 15) Should I be using Chan_SIP, PJSIP, or both on my server install. For hardware I have to support some old PolyCom 501 / 601’s which are being phased out. I have some Aastra 67xxi series phone which will be around for a few more years. For new phones looking seriously at the Sangoma phones.
So with this hardware, and future plans should I be using:
Asterisk 13 or Asterisk 15?
Chan_SIP or PJSIP or Both?
If there are consideration, in both the installation / configuration would be nice to know about those also.
As far as the Asterisk version to choose from, I would stick with the long term releases due to their longer life cycles vs the standard releases.
So I would use Asterisk 13 and upgrade to 16 next year, provided it will be a long term release which I assume is will be.
Been using Pjsip for 6 months with asterisk 13, it’s been pretty damn stable. More stable than chan_sip. I also find that I don’t have to reboot asterisk every few weeks to clear some stuck cache. You’ll just need to double-check and possibly recode any custom config you may have, as there are differences.
For sure use pjsip with your extensions. Chan_sip is dead and not receiving any kind of updates or bug fixes anymore from my understanding at Astricon in 2017
I must admit that I was convinced by all these posts here, that PJSIP is a no go if you don’t have much experience, while ChanSIP is rather plug-and-play.
Recently, I started throwing in a PJSIP extension here and there, I did now hear anyone complain or anyone noticing anything big of a difference.
Just two weeks ago we deployed our FIRST FreePBX with PJSIP Trunks and extentions. (of course I tested before moving into production).
I actually found that it’s easier for us to setup PJSIP Trunks than ChanSIP, and it works just fine.
One feature that I really relied on is, that you can always convert the extension to ChanSIP if necessary, and it’s not that you have to delete and recreate etc.
I can’t say I love PJSIP, since I didn’t yet explore all its features, but the multiple endpoints feature is great. (one downside, that you see “missed call” if you answered on another endpoint).
Digium supports chan_sip in Asterisk 13 so at least Digium is representing there will be at least security and bugfix support through 2021. I think it’s a little absolute/premature to say there’s “not any kind of updates” anymore.
Ok well at Astricon last year it was stated its getting only major security fixes no bug fixes. @tm1000 would know best but that is what I heard dunring Dev Con at Astricon last October.
Digium does not support chan_sip. Activity on the driver itself can be from external users but is reviewed/merged by a Digium developer and therefore will sometimes show up under their name
I just went out and asked (again). This message is directly from a Digium developer
Digium does not work on chan_sip, but if the community submits a change then we or others will review it
I asked about chan_sip last year at astricon as well. At astricon there are about 5 of the lead/head Digium developers. “Chan_sip” not supported comes directly from their mouths.
Just found this wiki write up from Astricon where they also say the same thing. Nobody at Digium mantiains it and at this time no outside community members were either.
What about stability? The FreePBX community has a large thread with users indicating issues with PJSIP that they don’t experience with chan_sip, but no one is filing bugs or presenting actual issues - it appears to be primarily anecdotal.
Is it already defacto deprecated since it’s in extended support and there is no community maintainer? And, are we being setup for something bad by not making it more clear.
We still have issues with PJSIP when we use TLS/SRTP
After a reload of freepbx all extensions are offline from time time and cannot register anymore until i reboot the pbx.
PJSIP solved a problem we had with people hacking our extensions to make international calls. Somehow sending random packets to Chan_SIP causes it to call out. It also connects faster and helps reloads, we have around 2000 extensions on the server. Works better for softphones too.
Would not go back to Chan_SIP its too unstable plus its dead. Developers have said there are major vulnerabilities but they will never be fixed nor talked about.
Because it’s categorized as extended, it’s not something for which Digium performs bug-fix maintenance. That doesn’t preclude anyone outside of Digium from performing that maintenance. In practice, however, that’s not very common. As it is, there’s not currently a named maintainer for chan_sip - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Open+Source+Maintainers
Since Asterisk 12, prior to 13 even, Digium’s focus has been on chan_pjsip, which we utilize in our commercial offering, Switchvox.
