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Hi Stas.<br>
<br>
A missing ACK may indicate a problem with the contact in the 200 OK
reply (contact pointing back to callee).<br>
Do you have a SIP capture on the OpenSIPS side ?<br>
<br>
Regards,<br>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 18.08.2016 16:40, Стас Тельнов
wrote:<br>
</div>
<blockquote
cite="mid:CAC89h=LbVaT18nh=cRvutCeJS5evBzK3kyVM9a0h909Y5-xLEA@mail.gmail.com"
type="cite">
<div dir="ltr">
<div dir="ltr">I have freeswitch and opensips working with the
mobile client in the conference mode.<br>
When using UDP connection everything works perfectly, but when
using tls connection the call is interrupted in 30 seconds.<br>
Whether to use TLS or UDP connection - it is assigned on the
mobile client before initialization of connection with
opensips server.<br>
<br>
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.<br>
<br>
Generally such scheme works as it should:<br>
<br>
+++++++++ udp ++++++++ udp +++++++++ udp +++++++++<br>
+ + -----> + + ----->
+ + -----> + +<br>
+ phone + + SIP + + free
+ + SIP +<br>
+ + <----- + + <----- +
switch + <----- + provider +<br>
+++++++++ udp ++++++++ udp +++++++++ udp
+++++++++<br>
<br>
And in such scheme a call breaks in 30 seconds:<br>
<br>
+++++++++ tls +++++++++ udp +++++++++ udp
+++++++++<br>
+ + -----> + + ----->
+ + -----> + +<br>
+ phone + + SIP + + free
+ + SIP +<br>
+ + <----- + + <----- +
switch + <----- + provider +<br>
+++++++++ tls +++++++++ udp +++++++++ udp
+++++++++<br>
<br>
SIP and freeswitch are in one local area network (Amazon EC2).
SIP provider doesn't support tls in principle, they have 5061
closed.<br>
<br>
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:<br>
BYE <a class="moz-txt-link-abbreviated" href="mailto:sip:8@85.236.*.*:55194;ob">sip:8@85.236.*.*:55194;ob</a>;<wbr>transport=tls SIP/2.0<br>
Via: SIP/2.0/TLS sip0.*.*:5061;branch=<wbr>z9hG4bKc7a2.7909e7e1.0;<wbr>received=52.58.*.*<br>
Via: SIP/2.0/UDP 172.31.*.*;received=52.58.*.*;<wbr>rport=5060;branch=<wbr>z9hG4bKBK82Zg50c2U0p<br>
Max-Forwards: 69<br>
Contact: <<a class="moz-txt-link-abbreviated" href="mailto:sip:*7906******@52.58.*.*">sip:*7906******@52.58.*.*</a>:<wbr>5060;transport=udp><br>
To: "8" <a class="moz-txt-link-rfc2396E" href="mailto:sip:8@sip0.*.*"><sip:8@sip0.*.*></a>;tag=59221e6a<br>
From: <a class="moz-txt-link-rfc2396E" href="mailto:sip:*7906******@sip0.*.*"><sip:*7906******@sip0.*.*></a>;<wbr>tag=j4aX21rv83etN<br>
Call-ID: O7E3ktwLPiQWDN2Rism-7g..<br>
CSeq: 95383912 BYE<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY<br>
Supported: timer, path, replaces<br>
User-Agent: FreeSWITCH-mod_sofia/1.6.6~<wbr>64bit<br>
Reason: SIP;cause=408;text="ACK Timeout"<br>
Content-Length: 0<br>
<br>
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.<br>
<br>
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?<br>
Who faced similar problem, prompt what settings should be
analyzed in order that the above-stated scheme with tls
connection start functionning?
<div class=""><br>
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<br>
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</pre>
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