NAT SIP phone to NAT Asterisk -- No audio?

I saw a post saying “open 10000-20000 for RTP” to Asterisk. I haven’t done that yet, but I wanted to ask the question.

I have an Aastra 9133i. It is at my house, behind a NAT device.

I have an Asterisk install at a friends house, also behind a NAT device.

The SIP phone is able to successfully register to Asterisk remotely, due to my nat=yes and such in the sip.conf.

I can call the SIP phone and have it ring.
I can dial an extension in the local context and have it work.

I cannot hear any audio on the SIP phone whatsoever.

Is opening port 10k-20k forwarded to Asterisk gonna resolve this? Do I have to open up anything on the NAT on the SIP phone side?

You need ports 10000 -> 20000 open and pointed to the asterisk server.

Not all SIP device are set to use the same range of ports…check what ports the phone is set to use.

and yes that is why you not getting audio.

en.wikipedia.org/wiki/Real-time_ … t_Protocol

The port range 10000 to 20000 are not magic numbers. The actual ports you need to open are the ones described in /etc/asterisk/rtp/conf by the “rtpstart” and “rtpend” parameters. The UDP ports you need to forward to your asterisk server are the ones described by that range. 10000-20000 are the default values, but check that is still how they are set. You do not actually need to use that many ports to run a home-sized asterisk.

Typo. I meant /etc/asterisk/rtp.conf.

asterisk (if behind nat) must have in sip.conf externip= and localnet= defined. It must also have port 5060 and (rtp range from rtp.conf) forwarded.

Phone (if behind nat) must have STUN enabled or by some other method know its external IP.

set nat=yes and canreinvite=no…