Hi,
when I receive a call in the "Contact" field is specified "transport=tls", instead of "udp".
In REGISTER the transport is UDP.
I tried disabling TLS in the external profile, but the result is the same.
Using TLS in "Contact" field, the caller doesn't send the ACK and the call ends.
Why does it use TLS instad of UDP?
Following parts of SIP packages:
when I receive a call in the "Contact" field is specified "transport=tls", instead of "udp".
In REGISTER the transport is UDP.
I tried disabling TLS in the external profile, but the result is the same.
Using TLS in "Contact" field, the caller doesn't send the ACK and the call ends.
Why does it use TLS instad of UDP?
Following parts of SIP packages:
Code:
35.35.35.35 = myprovider IP
20.20.20.20 = public fusionpbx IP
10.132.0.39 = internal fusionpbx IP
333333333333333 = caller ID
Code:
-----------
REGISTER: transport UDP is specified
2022/04/27 10:05:17.145861 10.132.0.39:5080 -> 35.35.35.35:5060
REGISTER sip:provider.com;transport=udp SIP/2.0
Via: SIP/2.0/UDP 20.20.20.20:5080;rport;branch=z9hG4bKtaZDv0t0167Km
Max-Forwards: 70
From: <sip:myuser@provider.com>;tag=K6vcyXZr5QvKN
To: <sip:myuser@provider.com>
Call-ID: 2c6c0324-14f0-41bc-ae14-7163f6251ced
CSeq: 50969566 REGISTER
Contact: <sip:gw+0ee4f60d-7b6e-44a5-b438-68e13cb851cc@20.20.20.20:5080;transport=udp;gw=0ee4f60d-7b6e-44a5-b438-68e13cb851cc>
Expires: 800
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0
-----------
Final OK: transport UDP is specified
2022/04/27 10:05:17.363408 35.35.35.35:5060 -> 10.132.0.39:5080
SIP/2.0 200 OK
From: <sip:myuser@provider.com>;tag=K6vcyXZr5QvKN
To: <sip:myuser@provider.com>;tag=91dc4c8-508a8c0-13c4-55013-119c88-539b78b4-119c88
Call-ID: 2c6c0324-14f0-41bc-ae14-7163f6251ced
CSeq: 50969567 REGISTER
User-Agent: RegistrarServer
Require: outbound
Allow: REGISTER
Contact: <sip:gw+0ee4f60d-7b6e-44a5-b438-68e13cb851cc@20.20.20.20:5080;transport=udp;gw=0ee4f60d-7b6e-44a5-b438-68e13cb851cc>;expires=799
Via: SIP/2.0/UDP 20.20.20.20:5080;rport=5080;branch=z9hG4bKUKr6XUB4yFy6F
Content-Length: 0
-----------
Incoming Call: In request URI UDP is specified
2022/04/27 10:05:34.214140 35.35.35.35:5060 -> 10.132.0.39:5080
INVITE sip:gw+0ee4f60d-7b6e-44a5-b438-68e13cb851cc@20.20.20.20:5080;transport=UDP SIP/2.0
From: "333333333333333"<sip:333333333333333@provider.com;user=phone>;tag=4a45418-508a8c0-13d3-55013-119c98-24007445-119c98
To: <sip:myuser@provider.com>
Call-ID: 4fe08e0-508a8c0-13d3-55013-119c98-54b62509-119c98
CSeq: 1 INVITE
Via: SIP/2.0/UDP 35.35.35.35:5060;branch=z9hG4bK-119c98-44cbb544-3057576a
Record-Route: <sip:35.35.35.35:5061;transport=TLS;lr>
Via: SIP/2.0/TCP 192.168.8.5:22800;branch=z9hG4bK-119c98-44cbb510-adfc6b3
Referred-By: "test_algho"<sip:test_algho@provider.com>
Max-Forwards: 69
Supported: join,replaces,timer,100rel
User-Agent: Softrunk 2.4.1.8
Allow-Events: hold,refer
Contact: <sip:GRUUMTQ1OTI2NjA0NzI0NjFlOTk2NTQ2MjVlMTAyM2JmMDQ=@provider.com>
Session-Expires: 3600
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO
Content-Type: application/sdp
Content-Length: 652
-----------
Session Progress: In contact field there is transport TLS, why?
2022/04/27 10:05:34.249894 10.132.0.39:5080 -> 35.35.35.35:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 35.35.35.35:5060;branch=z9hG4bK-119c98-44cbb544-3057576a
Via: SIP/2.0/TCP 192.168.8.5:22800;branch=z9hG4bK-119c98-44cbb510-adfc6b3
Record-Route: <sip:35.35.35.35:5061;transport=TLS;lr>
From: "333333333333333" <sip:333333333333333@provider.com;user=phone>;tag=4a45418-508a8c0-13d3-55013-119c98-24007445-119c98
To: <sip:myuser@provider.com>;tag=mFp5Zrgv20j6g
Call-ID: 4fe08e0-508a8c0-13d3-55013-119c98-54b62509-119c98
CSeq: 1 INVITE
Contact: <sip:myuser@20.20.20.20:5081;transport=tls>
User-Agent: FreeSWITCH
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 771
Remote-Party-ID: "myuser" <sip:myuser@provider.com>;party=calling;privacy=off;screen=no