<div dir="ltr">Yes, I've read this RFC.<br><br>In my case<span id="result_box" class="" lang="en"><span class=""><br></span></span><br><blockquote style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><span class="im"><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"><wbr>sport=tls</span></font><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif"> SIP/2.</span></div></div></span><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif">and</span><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif"></span><br><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif"></span><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"><wbr>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><br></div></div></blockquote><span class="im"><div dir="ltr"><div><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>is not an error.<br>In my case Alice (8@85.236. *. *) uses tls, but Bob (*7906 ****** @52.58. *. *) uses udp.<br><br>I can configure Alice softphone, freeswitch and opensips. Alice <--> uses only tls.<br>I can not configure sip of provider and Bob's phone. Sip provider <--> Bob uses only udp.<br><br>I believe that such scheme is possible. If I send BYE packet for example from Alice softphone, then<br>1. This packet goes on opensips using tls<br>2. opensips changes the protocol for udp and sends a packet to freeswitch.<br>3. freeswitch sends a packet using udp to sip of provider and it sends it to Bob.<br><br>Similarly when Bob sends BYE packet:<br>1. freeswitch accepts udp packet from provider and sends it to opensisps using udp<br>2. opensisps accepts this udp packet, changes the protocol for tls and sends this packet to Alice using tls.<br><br>But with ACK packet between sip provider and freeswitch something goes wrong.<br></span></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"></span></div></div></span><div class="gmail_extra"><br><div class="gmail_quote">2016-08-18 18:38 GMT+03:00 Sergey Safarov <span dir="ltr"><<a href="mailto:s.safarov@gmail.com" target="_blank">s.safarov@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">It may be root of issue<span class=""><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"><wbr>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></span><div dir="ltr"><div><span style="color:rgb(33,33,33);font-family:"helvetica neue",helvetica,arial,sans-serif">and<br></span></div></div><span class=""><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"><wbr>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></span><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 face="helvetica neue, helvetica, arial, sans-serif" color="#212121"><a href="https://tools.ietf.org/html/rfc3261#section-4" target="_blank">https://tools.ietf.org/<wbr>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><div class="h5"><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;<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: <sip:*7906******@52.58.*.*:<wbr>5060;transport=udp><br>To: "8" <sip:8@sip0.*.*>;tag=59221e6a<br>From: <sip:*7906******@sip0.*.*>;<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><br></div></div></div></div></div>
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