[asterisk-users] Asterisk pjsip and NAT just doesn't work
m1278468 at mailbox.org
Sun May 2 03:08:00 CDT 2021
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):
- 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
bind to 10.15.13.17 (may I bind to a interface name?)
What are the problems?
- 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.
- 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
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