<div dir="ltr">It may be root of issue<div dir="ltr"><div><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif">BYE sip:8@85.236.*.*:55194;ob;</span><font color="#ff0000"><span style="font-family:"helvetica neue",helvetica,arial,sans-serif">tran</span><span style="font-family:"helvetica neue",helvetica,arial,sans-serif">sport=tls</span></font><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif"> SIP/2.0</span><br></div><div><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif"><br></span></div></div><div dir="ltr"><div><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif">and<br></span></div></div><div dir="ltr"><div><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif">Contact: <sip:*7906******@52.58.*.*:506</span><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif">0;</span><span style="font-family:"helvetica neue",helvetica,arial,sans-serif"><font color="#ff0000">transport=udp</font></span><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif">></span><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif"><br></span></div></div><div dir="ltr"><div></div><div><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif"><br></span></div><div><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif">Look at </span><font color="#212121" face="helvetica neue, helvetica, arial, sans-serif"><a href="https://tools.ietf.org/html/rfc3261#section-4">https://tools.ietf.org/html/rfc3261#section-4</a> </font><span style="font-size:13.3333px;line-height:normal">Figure 1: SIP session setup example with SIP trapezoid</span></div><div><span style="font-size:13.3333px;line-height:normal"><br></span></div><div><span style="font-size:13.3333px;line-height:normal">Sergey</span></div><div><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif"><br></span></div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr">чт, 18 авг. 2016 г. в 18:28, Стас Тельнов <<a href="mailto:stasan89@gmail.com" target="_blank">stasan89@gmail.com</a>>:<br></div></div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><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 sip:8@85.236.*.*:55194;ob;transport=tls SIP/2.0<br>Via: SIP/2.0/TLS sip0.*.*:5061;branch=z9hG4bKc7a2.7909e7e1.0;received=52.58.*.*<br>Via: SIP/2.0/UDP 172.31.*.*;received=52.58.*.*;rport=5060;branch=z9hG4bKBK82Zg50c2U0p<br>Max-Forwards: 69<br>Contact: <sip:*7906******@52.58.*.*:5060;transport=udp><br>To: "8" <sip:8@sip0.*.*>;tag=59221e6a<br>From: <sip:*7906******@sip0.*.*>;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/1.6.6~64bit<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><br></div></div></div>
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