[asterisk-users] Asterisk as a client: can I get the remote SIP server to ignore rport?

Markus universe at truemetal.org
Thu Feb 20 19:12:57 CST 2014

Am 21.02.2014 01:33, schrieb Eric Wieling:
> To be fair NAT is rewriting your SIP packet source port.  This happens all day, on almost every NAT device out there.    Stop thinking it is purely a port rewriting issue, something else is going on.

In the meantime, the provider has reconfigured the VM to work with the 
public IP address. That means the RFC IP address was removed and the 
public IP is now configured on the VM directly. The effect is the same, 
ports on outgoing packets still get rewritten.

> Have you set localnet and externip in sip.conf.  Maybe the NAT device has a short UDP translation timeout -- try setting qualifyfreq=15 in sip.conf to generate traffic so the NAT box does not close the translations

Yes, I have played around with local and externaddr. But the IP is not 
the problem, the port is. I think the translation timeout doesn't 
matter, because the router rewrites outgoing packets to a different 
port, but doesn't do so on incoming packets, and that's the issue.

> Here is an example sip show peers on one my my boxes.   Three different locations are show.  The ones you see with 5060 are either not NAT'd or they have a proxy at the customer location.  The ones with a different port are NAT'd.

And I'm pretty sure if you look at any of those peers that have a 
non-5060 port, the routers in front of them will rewrite packets 
destined for ports 53277, 4121, 47822 etc. to the proper corresponding 
internal IP:port where something is listening. The router of my provider 
won't. It rewrites ports on outgoing packets, but it passes incoming 
packets 1:1 to the VM.

IMHO, my hosting provider is at fault, and I'm working with them to get 
it fixed. I was just wondering if there is some magic switch which can 
fix such a broken scenario.

Thank you!

More information about the asterisk-users mailing list