General Advice: moving from 3CX

[edwindrood]

I am looking to move away from 3CX to FusionPBX on a VPS like OVH. I have seen lots of great tutorials.
What I am having troubles with is understanding best practices for desk-phones in the office. I have mostly Polycom VVX 410 (these are still cheap on ebay).
Of course, 3CX always uses a proprietary SBC in each remote office.

Although the vvx 410 version 5.9.x is supposed to have STUN support, I do not see it. Plus I see many folks here advising to not use STUN at all. If all remote offices have a static outside IP address, I see there may be options for that, but I would prefer the ability to support a dynamic ISP.

I see that I could possibly set up an equivalent proxy as mentioned here:

SBC_Setup | FreeSWITCH Documentation

About

developer.signalwire.com

Even better, I see from the docs that I could possibly both provision and set up SIP routing like this:

Common Use Cases — dSIPRouter documentation

I would prefer to not toss all my Polycom vvx 410 phones into the landfill.
I would prefer a light-weight SBC (or SIP Proxy) in each remote office if I am unable to connect another way.

I do not want to re-invent the wheel. I am sure that there are many folks who have already addressed this.

I am not ready to dip my toe into multi-tenant stuff yet. I am just looking at supporting a few phones in a few locations.

So please let me know: Is having a light-weight Proxy in each remote location the best way to go?

 

[hfoster]

I have personally never needed to use STUN/ICE, etc. I only rely on the rport mechanism of SIP and FreeSWITCH/mod_sofia's ability to deal with NAT on the client sites. The phones are behind a mix of consumer routers and some enterprise routers, Cisco, Juniper, Drayteks, Sonicwall etc.

I believe 3CX has such a push for it, because they don't want their partners overwhelmed with people who configure absolutely crazy firewalls that just block arbitrary ports outbound, apply all sorts of crazy ALG manipulations and other nonsense.

Edit: Updated wording to highlight that rport is how I do NAT traversal.

 

[pbxgeek]

I use a ton of older VVX410 on 5.9.8 firmware, and I don't do any STUN. Some clients have static IPs, and some have Dynamic IPs. There was an issue with SIP ALG with some routers and modems some time ago, before I implemented TLS. Since then, it's been pretty much plug-and-play in any office with any router. If you want, give me a call, and I'm happy to walk you through how I set up those Polycoms.

 

[RTL]

No need for a proxy. 3cx only use it because it doesn't traverse nat well. Fusionpbx/freeswitch however does it well. We have a lot of vvx 410 handsets and we just provision with the template provided. Don't overthink this if you come from 3cx, they overcomplicate the nat issue