SIP ALG, FreeSwitch, and Virtual Contact Centers

random_nerd

New Member
Sep 30, 2025
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Hi all! We just migrated to FreeSwitch to cut costs, gain control, etc.

Since the switch, however, I'm hearing many more agent complaints surrounding dropped call connection.

We run a virtual contact center and, when checking agent's configurations, most/all have SIP ALG enabled by default on their routers... and most have ISP's that either don't know what it is, or have reps who refuse/decline the ability to switch it off.

With or previous dialer (Five9) this didn't seem to be a big of an issue, but now it seems like a lot of dead air, one way audio... just issues in general.

Do you guys have any tips for managing this? (Assuming SIP ALG Cann't be switched off) Maybe we could route/hide all their call traffic through a VPN? Is FS particularly sensitive to this sort of thing? Maybe we have to go multi-box? (We're single-boxed on an AWS EC2 machine)... trying trying to get some direction here.

Thank you!
 
Your answer is SIP TLS. It’s true and tested. It will bypass any SIP ALG.
Derp. I was just reading into that, and I see it's not currently enabled on my inbound or outbound profiles... Is this relatively easy to set up? Agents connect to FS via our web app/sip js.
 
Turns out, I DID have it installed, and I was already connecting to my sip endpoint via wss, so signaling is probably fine... I guess. And I guess this means it is outside the purview of SIP ALG. Would that mean if agents are experiencing dropped calls it would have to be an RTP/Audio transport issue? I'm reading SRTP would be the solution for that? And then between SIP TLS + SRTP I would probably be fully shielded from SIP ALG issues? Seeing the only other option after that would be a TURN server but that seems wildly overcomplicated for what we're doing here.
Should be easy.