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.