[asterisk-users] problems with natted phones

Marek GreŇ°ko mgresko8 at gmail.com
Thu Jul 8 13:57:30 CDT 2021


Hello,

I have an asterisk setup using pjsip. Everything used to work
correctly until one remote site changed internet provider and thier
router does not support sip protocol algorithms.

It works for some time, but then suddenly audio stops working both
directions. When this happens I see RTP responses going out to the
local address of the natted phone, not to the natted address. The
problem appears for the phones independently.

The asterisk is connected to the internet with public static IP address.

The pjsip config contains:

[aor]
type=aor
qualify_frequency = 60
max_contacts=1
remove_existing = yes

[endpoint]
type = endpoint
context = internal
dtmf_mode = rfc4733
disallow = all
allow = alaw
allow = ilbc
allow = g729
allow = gsm
allow = g723
direct_media = no
allow_subscribe = yes
subscribe_context = blf
rewrite_contact = yes
rtp_symmetric = yes
force_rport = yes


Am I missing something? Why the communication breaks suddenly?

Thanks

Marek



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