Not Audio PJSIP 2.5 and Asterisk 14 - NAT

Hi,

not audio in call asterisk 14 my conf:

rtp.conf
icesupport=true
stunaddr:19302=stun.l.google.com

pjsip.conf
[transport-udp]
type = transport
protocol = udp
bind = 192.168.10.218:5060
local_net=192.168.0.0/16
external_media_address=201.190.16.62
external_signaling_address=201.190.16.62

My endpoint enable options:
[6001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
transport=transport-udp
auth=6001
aors=6001
direct_media=no
force_rport=yes
rtp_symmetric=yes
ice_support=yes
rwrite_contact=yes

[6001]
type=auth
auth_type=userpass
password=*********
username=6001

[6001]
type=aor
max_contacts=2

Asterisk server Connections
192.168.10.218:5060 <- 201.190.16.62:55000(nat 55000 to 5060) <- Internet

Endpoint Connections
192.169.40.233 -> 177.79.78.228(dynamic) -> Internet

Cli pjsip show contacts
Contact: 6001/sip:5361@177.79.78.228:5060;transport=UDP 4a95161c75 Unknown nan

Cli rtp set debug :sob:
Sent RTP packet to 192.169.40.233:8000 (type 03, seq 037093, ts 12481101, len 000033)

You’ll need to provide the SIP signaling and console output, to ensure that the SDP provided to the client contains the correct address. If it does then it’s outside of Asterisk.