Call Drop in TLS transport

Asterisk Asterisk SIP
1

Hi,

I configured Asterisk 16 to use TLS as transport protocol.
The calls were working good with 2-way audio, but drops after 32 seconds.

The configurations used :

##pjsip.conf

[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0
cert_file=/etc/letsencrypt/live/Domain/fullchain.pem
priv_key_file=/etc/letsencrypt/live/Domain/privkey.pem

##endpoint

[1900]
type=aor
max_contacts=1
remove_existing=yes

[1900]
type=auth
auth_type=userpass
username=1900
password=password
[1900]
type=endpoint
aors=1900
auth=1900
use_avpf=yes
ice_support=yes
media_use_received_transport=yes
context=context
disallow=all
allow=opus,alaw
allow=vp8,h264
identify_by=username,auth_username
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
dtmf_mode=rfc4733
send_rpid=no
trust_id_inbound=yes

I am getting this warning :

WARNING[12802]: pjproject: <?>: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <167773202> <SSL routines-???-sslv3 alert bad certificate> len: 0 peer: DEVICE_IP:49834

Please help me to resolve this issue.

2

Are you behind NAT?

3

yes

4

Your transport has not been configured with the local_net, external_media_address, and external_signaling_address options so the remote side is most likely trying to connect to the wrong place to send the ACK.

5

Thanks

That helps

6

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