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

[Bogdan-Andrei Iancu]

Hi Stas.

A missing ACK may indicate a problem with the contact in the 200 OK 
reply (contact pointing back to callee).
Do you have a SIP capture on the OpenSIPS side ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.08.2016 16:40, Стас Тельнов wrote:
> 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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