[asterisk-users] PJSIP, NAT and STUN/ICE

O. Hartmann ohartmann at walstatt.org
Mon Oct 9 16:56:17 CDT 2017

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a
NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed
to act as the telephone gateway for several VoIP/SIP phones.

I'm using throughout pjsip as configuration, I have no experience with chan_sip since I
started recently using Asterisk for several SoHo and lab's projects. So be alarmed, there
may come some noobish questions.

When planning and setting up the Asterisk, I had to deal with NAT. The Asterisk config
object of type=transport knows about essential entries:

external_media_address=		dyndns FQDN
external_signaling_address=	dyndns FQDN
direct_media=			no
rtp_symmetric=			yes
force_rport=			yes

dyndns FQDN is the FQDN of my broadband access point provided by some dynamical DNS

This setup is not working properly with when external_media_address= and
external_signaling_address= are set that way, but commenting out both makes all of the
ITSP which provides me with service happy.

I think, at this point I have no idea of the concept or, there is simply a missing link,
even a (dangerously) misconfigured router - on which the Asterisk runs. When
external_xxxx_xxxx are not set, I suppose they're set to, aren't they,
implying that they listen on all configured IPs?

If so, is there a way to show the setting of external_media_address= and
external_signaling_address= on the CLI?

When using PJSIP with the setting excluding attributes external_xxx_xxx, does pjsip do
some magic to traverse the NAT in a proper way? At this very moment, I do not understand
how things would work avoiding setting external_media_address= and
external_signaling_address=. The Asterisk 13 I'm running is supposed to be bound to IP, which is routed to the gateway. 

My problem is, I don not understand why the communication is working fluently with both
external_xxx_xxx atrributes commeneted out which I suppose to be crucial - following the
Asterisk 13 documentation!

Maybe someone can sched some light onto this. I think, my view on the matter is
completely confused.

Thanks in advance,


Another point
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