[asterisk-users] Asterisk pjsip and NAT just doesn't work

[Michael Maier]

Hello!

I've just playing around some time to get NAT and pjsip running with asterisk 18.3 
and 18.4 (w/o any patches added). NAT should be used for connection to the trunk.

I wasn't able to get it working, because SDP address rewriting just doesn't work 
as it should.

The situation is like this (CentOS 7):

- Multihomed
- One small net for the trunk 10.15.13.17/32 as alias on one of the existing devices.
- TLS for SIP
- Added complete masquerading for this IP address
- added dnsmanager which provides the global IP address
- used direct media no
- used rewrite contact yes or no
- force rport: yes
- transport:
	external_media_address=external.mydom.org
	external_signaling_address=external.mydom.org
	bind to 10.15.13.17 (may I bind to a interface name?)


What are the problems?

Outbound calls:
- Biggest problem: even if the WAN IP is set everywhere correctly in the *initial* 
INVITE, it's *always* missing in the INVITE *after* the 407 Proxy auth request in 
the SDP. In the first Invite, the SDP was ok, in the second Invite, the SDP is 
broken (rewriting doesn't seem to happen). Such calls naturally are dropped by the 
ISP (ok, one of my providers seems to ignore the entry completely).

- Another problem is, that the given external IP just isn't used consistently, 
some times it's there - mostly not (always the same easy call setup). I suspect / 
fear different behavior between reload and restart with same configuration.

- I expect all IP addresses of mine in all sip headers have to be the WAN IP.



Inbound calls:
- Playing announcements doesn't work at all (no sound though rtp packages are 
flowing in both directions according tcpdump at the WAN interface).
- Calls given to local devices are working.



Could somebody maybe give me a reference configuration for a working NAT 
configuration?


Thanks
Michael