Retransmission timeout reached on transmission with SSL

Hiii Asetrisk team,
Please once I enable SSL on asterisk the call drop after 2 sec with this log:
[2019-08-31 10:15:48] VERBOSE[1936] asterisk.c: Remote UNIX connection
[2019-08-31 10:16:13] VERBOSE[6353] chan_sip.c: Registered SIP ‘BBE357NDM’ at XXX.XXX.XXX.XXX:58422
[2019-08-31 10:16:13] VERBOSE[6353] chan_sip.c: Saved useragent “Makane Android 2.9” for peer BBE357NDM
[2019-08-31 10:16:18] VERBOSE[6353][C-00000009] netsock2.c: Using SIP RTP TOS bits 184
[2019-08-31 10:16:18] VERBOSE[6353][C-00000009] netsock2.c: Using SIP RTP CoS mark 5
[2019-08-31 10:16:18] VERBOSE[6356][C-00000009] pbx_realtime.c: Executing [OZC391JGH@from-sip:1] Gosub(“SIP/BBE357NDM-00000012”, “chexec,1(OZC391JGH)”)
[2019-08-31 10:16:18] VERBOSE[6356][C-00000009] pbx_realtime.c: Executing [chexec@from-sip:1] Set(“SIP/BBE357NDM-00000012”, “TMOUT=60”)
[2019-08-31 10:16:18] VERBOSE[6356][C-00000009] pbx_realtime.c: Executing [chexec@from-sip:2] Dial(“SIP/BBE357NDM-00000012”, “SIP/OZC391JGH”)
[2019-08-31 10:16:18] VERBOSE[6356][C-00000009] netsock2.c: Using SIP RTP TOS bits 184
[2019-08-31 10:16:18] VERBOSE[6356][C-00000009] netsock2.c: Using SIP RTP CoS mark 5
[2019-08-31 10:16:18] VERBOSE[6356][C-00000009] app_dial.c: Called SIP/OZC391JGH
[2019-08-31 10:16:19] VERBOSE[6356][C-00000009] app_dial.c: SIP/OZC391JGH-00000013 is ringing
[2019-08-31 10:16:23] VERBOSE[6356][C-00000009] app_dial.c: SIP/OZC391JGH-00000013 answered SIP/BBE357NDM-00000012
[2019-08-31 10:16:23] VERBOSE[6357][C-00000009] bridge_channel.c: Channel SIP/OZC391JGH-00000013 joined ‘simple_bridge’ basic-bridge <5882e79f-14dc-43fa-a57d-7d5c092e1114>
[2019-08-31 10:16:23] VERBOSE[6356][C-00000009] bridge_channel.c: Channel SIP/BBE357NDM-00000012 joined ‘simple_bridge’ basic-bridge <5882e79f-14dc-43fa-a57d-7d5c092e1114>
[2019-08-31 10:16:26] WARNING[2483] chan_sip.c: Retransmission timeout reached on transmission 3dc99a8d-608f-47d2-91cd-3af220d1661c for seqno 31674 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 3199ms with no response
[2019-08-31 10:16:26] WARNING[2483] chan_sip.c: Hanging up call 3dc99a8d-608f-47d2-91cd-3af220d1661c - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2019-08-31 10:16:26] VERBOSE[6356][C-00000009] bridge_channel.c: Channel SIP/BBE357NDM-00000012 left ‘simple_bridge’ basic-bridge <5882e79f-14dc-43fa-a57d-7d5c092e1114>
[2019-08-31 10:16:26] VERBOSE[6356][C-00000009] pbx.c: Spawn extension (from-sip, chexec, 2) exited non-zero on ‘SIP/BBE357NDM-00000012’
[2019-08-31 10:16:26] VERBOSE[6357][C-00000009] bridge_channel.c: Channel SIP/OZC391JGH-00000013 left ‘simple_bridge’ basic-bridge <5882e79f-14dc-43fa-a57d-7d5c092e1114>

Please note that using udp protocol , everything is working good.
Best regards

You should provide an actual SIP trace using “sip set debug on” to show the contents and what is going on.