Not Audio PJSIP 2.5 and Asterisk 14 - NAT

Asterisk Asterisk SIP
1

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)

2

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.