No audio using SIP proxy and TLS


<--- Transmitting SIP request (1412 bytes) to TLS:157.230.97.157:39297 --->
INVITE sips:92201@157.230.97.157:39297;transport=TLS;fs-conn-id=f5603645b34c58f1;CtRt2ca863acf1b49e77=tls:80.17.99.73:1677 SIP/2.0
Via: SIP/2.0/TLS 157.230.97.157:5061;rport;branch=z9hG4bKPj8d035541-42b9-4844-b377-0f62577e0e5f;alias
From: "3xxxxxxxxxx" <sip:3xxxxxxxxxx@157.230.97.157>;tag=703768ab-c8d5-44c3-abdd-ac96f64bfaf8
To: <sips:92201@157.230.97.157;fs-conn-id=f5603645b34c58f1;CtRt2ca863acf1b49e77=tls:80.17.99.73:1677>
Contact: <sips:asterisk@157.230.97.157:5061;transport=TLS>
Call-ID: 803b404e-937e-413b-9826-3900f99ffa9e
CSeq: 3396 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "3xxxxxxxxxx" <sip:3xxxxxxxxxx@157.230.97.157>
Alert-Info: <http://www.notused >\;info=ring2
Max-Forwards: 70
User-Agent: FPBX-14.0.13.12(13.34.0)
Content-Type: application/sdp
Content-Length:   435

v=0
o=- 562606718 562606718 IN IP4 157.230.97.157
s=Asterisk
c=IN IP4 157.230.97.157
t=0 0
m=audio 16900 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 17426 RTP/AVP 99 104
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=sendrecv


<--- Received SIP response (675 bytes) from TLS:157.230.97.157:39297 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 157.230.97.157:5061;rport=5061;branch=z9hG4bKPj8d035541-42b9-4844-b377-0f62577e0e5f;alias
Record-Route: <sips:157.230.97.157:6061;lr>
From: "3xxxxxxxxxx" <sip:3xxxxxxxxxx@157.230.97.157>;tag=703768ab-c8d5-44c3-abdd-ac96f64bfaf8
To: <sips:92201@157.230.97.157;fs-conn-id=f5603645b34c58f1;CtRt2ca863acf1b49e77=tls:80.17.99.73:1677>;tag=2621131302
Call-ID: 803b404e-937e-413b-9826-3900f99ffa9e
CSeq: 3396 INVITE
Contact: <sip:92201@80.17.99.73:1677;transport=tls>
User-Agent: Fanvil X4U 1.0.0 0c383e4249ef
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (1088 bytes) from TLS:157.230.97.157:39297 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 157.230.97.157:5061;rport=5061;branch=z9hG4bKPj8d035541-42b9-4844-b377-0f62577e0e5f;alias
Record-Route: <sips:157.230.97.157:6061;lr>
From: "3xxxxxxxxxx" <sip:3xxxxxxxxxx@157.230.97.157>;tag=703768ab-c8d5-44c3-abdd-ac96f64bfaf8
To: <sips:92201@157.230.97.157;fs-conn-id=f5603645b34c58f1;CtRt2ca863acf1b49e77=tls:80.17.99.73:1677>;tag=2621131302
Call-ID: 803b404e-937e-413b-9826-3900f99ffa9e
CSeq: 3396 INVITE
Contact: <sip:92201@80.17.99.73:1677;transport=tls;verified>
Supported: 100rel, replaces, timer
User-Agent: Fanvil X4U 1.0.0 0c383e4249ef
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 360

v=0
o=92201 293661303 6030219836 IN IP4 80.17.99.73
s=A conversation
c=IN IP4 157.230.97.157
t=0 0
m=audio 25356 RTP/AVP 0 8 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
m=video 0 RTP/AVP 99
a=rtpmap:99 H264/90000

<--- Transmitting SIP request (596 bytes) to TLS:157.230.97.157:39297 --->
BYE sip:92201@80.17.99.73:1677;transport=tls SIP/2.0
Via: SIP/2.0/TLS 157.230.97.157:5061;rport;branch=z9hG4bKPjafe72b39-cef2-48e4-936c-a0789e0c5f46;alias
From: "3xxxxxxxxxx" <sip:3xxxxxxxxxx@157.230.97.157>;tag=703768ab-c8d5-44c3-abdd-ac96f64bfaf8
To: <sips:92201@157.230.97.157;fs-conn-id=f5603645b34c58f1;CtRt2ca863acf1b49e77=tls:80.17.99.73:1677>;tag=2621131302
Call-ID: 803b404e-937e-413b-9826-3900f99ffa9e
CSeq: 3397 BYE
Route: <sips:157.230.97.157:39297;transport=TLS;lr>
Warning: 381 SIP "SIPS Required"
Max-Forwards: 70
User-Agent: FPBX-14.0.13.12(13.34.0)
Content-Length:  0