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:[email protected].*.*: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@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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