Configuring SIP

[viktoriyanavrotskaya]

@irvinewade

Continuing the discussion from Question About Status of PostmarketOS on Librem 5:

@SteveR

Configuring that Grandstream box is like configuring a router, except much more difficult.

I hear you, bro.

One reason that it is “much more difficult” is that not only do the settings in the VoIP box have to be correct and the settings in the VoIP account have to be correct but … because VoIP is a somewhat more advanced network protocol, it has to work hard just to get through the router, which among other things may be doing NAT. That is to say, the settings on the router have to be correct too. (And technically if your ISP is “Big Brother”, the ISP itself may interfere with correct operation. This might particularly apply where you are attempting to use mobile data as your ISP and your mobile network operator isn’t really expecting customers to be using SIP. So in that case there is no router as such but the mobile network operator is causing some of the same problems.)

Yeah, NAT is usually the real culprit with SIP. The classic gotchas are one-way audio or calls dropping after 30s, which almost always trace back to RTP ports not getting back through the router. Worth checking whether your Grandstream has a STUN server set and whether SIP ALG is switched off on the router, since ALG “help” tends to mangle things more than fix them. Static internal IP for the box helps too so port forwarding stays put.

Yep, that dropping-after-30s symptom is the classic RTP-not-getting-back-through tell, so it’s almost certainly NAT and not the provider. I’ll check the Grandstream’s STUN setting and, more importantly, kill SIP ALG on the router, that “help” mangles headers way more than it ever fixes. And good call on pinning a static internal IP for the box so the port forwards don’t drift when the lease renews. That combo usually clears the one-way audio too. Thanks, going to run through these now.