PJSIP Follow Me No Audio

[lgaetz (Lorne Gaetz)]

@alexcal

I opened ports 10000 to 20000 on my router and this fixed

This is one of the cases where forwarding the RTP port range is often required, as I’ve described in detail here:

SIP Port Forwarding

The nuts and bolts of SIP are complicated, but put simply: SIP session negotiation takes place over the signalling port (default 5060) and the audio (more correctly, the ‘media’) goes over a random pair of ports in the RTP port range (default 10k-20k). It’s quite common for providers to do SIP signalling and SIP media from different hosts, and some providers have many media servers. So imagine what happens when a SIP session is negotiated through a NAT router, but the audio appears from a differ…