Hello. please could somebody help to determine cause of issue.
Asterisk version is 16.18.0
I have voice delay(3-10 seconds). And tried to used following configuration of pjsip endpoints: using autogenetation certs
[webrtc-endpoint-template](!)
type = endpoint
context = from-internal
disallow = all
allow = opus
webrtc = yes
rtp_symmetric = yes
rewrite_contact = yes
force_rport = yes
and cert file with key
[webrtc-endpoint-template](!)
type=endpoint
context=from-internal
disallow=all
allow=opus
;direct_media=no
media_encryption=dtls
dtls_verify=fingerprint
dtls_auto_generate_cert=no
dtls_cert_file=/path/cert.pem
dtls_private_key=/patch/secret.key
dtls_setup=actpass
use_avpf=yes
ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes
But I have delay when asterisk send first Hello packet. As usual that delay is 3-10 seconds.
on wireshark screen is dump from client. Usually I’ve got dump from server and looks like Asterisk send first Hello DTLS packages with random delay.
Please could you focus what’s can I check else?
Client negotiate ICE immediately according log and STUN interconnections.