[OpenSIPS-Users] Configure freeswitch and opensips for using tls and udp protocols simultaneously.

[Стас Тельнов]

I have freeswitch and opensips working with the mobile client in the
conference mode.
When using UDP connection everything works perfectly, but when using tls
connection the call is interrupted in 30 seconds.
Whether to use TLS or UDP connection - it is assigned on the mobile client
before initialization of connection with opensips server.

Originally I assumed that these problems were caused by the NAT settings,
but in that case the problem would be watched irrespective of the
connection used - UDP or TLS.

Generally such scheme works as it should:

+++++++++   udp   ++++++++   udp   +++++++++   udp   +++++++++
+               + ----->  +              +  ----->  +               +
----->  +               +
+   phone  +           +   SIP     +             +    free    +           +
    SIP    +
+               + <-----  +              +  <-----  +   switch  + <-----  +
provider +
+++++++++   udp   ++++++++   udp    +++++++++   udp   +++++++++

And in such scheme a call breaks in 30 seconds:

+++++++++   tls   +++++++++   udp   +++++++++   udp   +++++++++
+               + ----->  +               +  ----->  +               +
----->  +               +
+   phone  +           +   SIP      +             +    free    +
+     SIP    +
+               + <-----  +               +  <-----  +   switch  + <-----
+ provider +
+++++++++   tls   +++++++++   udp    +++++++++   udp   +++++++++

SIP and freeswitch are in one local area network (Amazon EC2). SIP provider
doesn't support tls in principle, they have 5061 closed.

And the BYE packet sends freeswitch, as I understand, from packet headers
as I didn't receive the response to ACK in time. There is the packet:
BYE sip:8 at 85.236.*.*:55194;ob;transport=tls SIP/2.0
Via: SIP/2.0/TLS sip0.*.*:5061;branch=z9hG4bKc7a2.7909e7e1.0;
received=52.58.*.*
Via: SIP/2.0/UDP 172.31.*.*;received=52.58.*.*;rport=5060;branch=
z9hG4bKBK82Zg50c2U0p
Max-Forwards: 69
Contact: <sip:*7906******@52.58.*.*:5060;transport=udp>
To: "8" <sip:8 at sip0.*.*>;tag=59221e6a
From: <sip:*7906******@sip0.*.*>;tag=j4aX21rv83etN
Call-ID: O7E3ktwLPiQWDN2Rism-7g..
CSeq: 95383912 BYE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
Reason: SIP;cause=408;text="ACK Timeout"
Content-Length: 0

Having looked on logs, I can tell that the INVITE packet from the mobile
client reach freeswitch and provider, but in reverse Trying/Ringing packet
doesn't reach.

I can't understand at what stage there is a problem. Freeswitch can't
respond and transmit the response through opensips, or there is a problem
in something else?
Who faced similar problem, prompt what settings should be analyzed in order
that the above-stated scheme with tls connection start functionning?
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