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

Guido Falsi mad at madpilot.net
Tue Oct 10 02:32:54 CDT 2017

On 10/09/2017 23:56, O. Hartmann wrote:
> 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:
> local_net=            
> bind=                 
> 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
> provider.
> 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?

the external_xxx settings are not configuring where to listen. Simply
tell asterisk what IP to put in the SDP data. You don't need to suppose
or guess the content of the SIP/SDP packets, you can look at them by
enabling asterisk debug output.

you can enable it following this guide:


Guido Falsi <mad at madpilot.net>

More information about the asterisk-users mailing list