<div dir="auto">Adding a stun can help too, if it is Linphone on Android then please add <a href="http://stun.linphone.org">stun.linphone.org</a> in stun settings further disable AVFP & ICE</div><br><div class="gmail_quote"><div dir="ltr">On Sat, Feb 2, 2019, 9:32 AM Alexey Sibyakin <<a href="mailto:alex@freeswitch.com">alex@freeswitch.com</a> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div><br></div><div>Yes, that sounds like a network (NAT/Firewall) issue. Your mobile provider may use a chain of several NATs and not every one of them is working properly. Try VPN or another SIP client.</div><div><br></div><div>Regards,</div><div><br></div><div>Alex</div><div><br></div><div><div><div dir="ltr" class="m_3830036265585201112m_-592515847464543468gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div><div dir="ltr" style="line-height:1.38;margin-top:0pt;margin-bottom:0pt;margin-left:4.5pt"><font color="#000000" face="arial, helvetica, sans-serif" size="1"><span style="white-space:pre-wrap">Alex Sibyakin | Support Engineer
SignalWire | 228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303
Email: <a href="mailto:alex@freeswitch.com" target="_blank" rel="noreferrer">alex@freeswitch.com</a>
Website: <a href="https://www.signalwire.com" target="_blank" rel="noreferrer">https://www.signalwire.com</a></span></font><br></div><div dir="ltr" style="line-height:1.38;margin-top:0pt;margin-bottom:0pt;margin-left:4.5pt"><font color="#000000" style="font-family:HelveticaNeue,"Helvetica Neue",Helvetica,Arial,"Lucida Grande",sans-serif;color:rgb(0,0,0);font-size:12.8px"><br></font></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Sat, Feb 2, 2019 at 2:44 AM Chhorm Chhatra <<a href="mailto:ch.chhatra@gmail.com" target="_blank" rel="noreferrer">ch.chhatra@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">Hi,<br><div>Currently, I face an issue that I could not hear the audio sound at all when I try to call to FreeSWITCH (9386) via its public domain name from the SIP client of my mobile phone with mobile internet.</div><div>In this case, SIP registration with TLS and media encryption with SRTP is working perfectly because I could see the secure symbol in Linphone <b>but</b> I could not hear any audio at all. After 30 seconds of the silent call, it hanged up automatically.</div><div><br></div><div><i>SIP Client on the 3G Mobile Internet > Freeswitch (tls:<a href="http://voip.test.com" target="_blank" rel="noreferrer">voip.test.com</a>, SRTP) </i></div><div><i>SIP Registration Sucess, Call No Media</i></div><div><br></div><div>My question is that is this the problem of NAT? If so, it would be great if you could help me solve this problem. </div><div>Any help would be really appreciated.</div><div><br></div><div>Best regards,</div><div>Chhorm Chhatra</div></div>
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