<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><meta http-equiv=Content-Type content="text/html; charset=utf-8"><meta name=Generator content="Microsoft Word 15 (filtered medium)"><style><!--
/* Font Definitions */
@font-face
{font-family:"MS Mincho";
panose-1:2 2 6 9 4 2 5 8 3 4;}
@font-face
{font-family:"MS Mincho";
panose-1:2 2 6 9 4 2 5 8 3 4;}
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:"\@MS Mincho";
panose-1:2 2 6 9 4 2 5 8 3 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0cm;
margin-bottom:.0001pt;
font-size:11.0pt;
font-family:"Calibri",sans-serif;}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:#0563C1;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:#954F72;
text-decoration:underline;}
span.EmailStyle17
{mso-style-type:personal;
font-family:"Calibri",sans-serif;
color:windowtext;}
span.EmailStyle18
{mso-style-type:personal;
font-family:"Calibri",sans-serif;
color:windowtext;}
span.EmailStyle19
{mso-style-type:personal-reply;
font-family:"Calibri",sans-serif;
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-size:10.0pt;}
@page WordSection1
{size:612.0pt 792.0pt;
margin:72.0pt 72.0pt 72.0pt 72.0pt;}
div.WordSection1
{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-AU link="#0563C1" vlink="#954F72"><div class=WordSection1><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>Thankyou for your reply. It got me thinking for a few days and a lot of testing.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>It looks very much like the router at the remote phone has some form of Sip ALG.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>Finding with that brand in general, having a reputation for locked enabled Sip ALG.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>The change is very subtle, easily glossed over and so it took some understanding.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>I have all the ports sorted out correctly now, so all that is all working properly.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>On Wireshark for inbound calls the rtp arrives correctly and actually has audio on it<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>but the phone itself does not reproduce the audio at all and it is totally silent and<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>there is no local outgoing rtp stream at all. Outbound calls now all work correctly.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>I even swapped phones over and once again had exactly the same problem.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>There is no problem at all when we use direct IP to IP calling through this router.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>So, this is somewhat of a larger problem, I am now looking at implementing TLS.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>Alex<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US>From:</span></b><span lang=EN-US> Users [mailto:users-bounces@lists.opensips.org] <b>On Behalf Of </b>Newlin, Ben<br><b>Sent:</b> Friday, 14 April 2017 10:04 PM<br><b>To:</b> OpenSIPS users mailling list<br><b>Subject:</b> Re: [OpenSIPS-Users] register phone in same local network as opensips<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-US>It doesn’t sound like it has anything to do with the registration. It sounds like your router has some sort of SIP Helper application that is trying to assist by re-writing the ports in the INVITE. Many modern routers come with this functionality enabled by default, even though in my experience it does nothing but break SIP communications.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Take a look at your router documentation for any mention of SIP functionality and disable it.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><div><p class=MsoNormal><b><span lang=EN-US style='font-size:10.5pt;color:black'>Ben Newlin </span></b><u><span lang=EN-US style='font-size:10.5pt;color:black'><o:p></o:p></span></u></p></div><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US style='color:black'>From: </span></b><span lang=EN-US style='color:black'>Users <</span><a href="mailto:users-bounces@lists.opensips.org"><span lang=EN-US>users-bounces@lists.opensips.org</span></a><span lang=EN-US style='color:black'>> on behalf of Alexander Jankowsky <</span><a href="mailto:E75A4669@exemail.com.au"><span lang=EN-US>E75A4669@exemail.com.au</span></a><span lang=EN-US style='color:black'>><br><b>Reply-To: </b>OpenSIPS users mailling list <</span><a href="mailto:users@lists.opensips.org"><span lang=EN-US>users@lists.opensips.org</span></a><span lang=EN-US style='color:black'>><br><b>Date: </b>Friday, April 14, 2017 at 8:08 AM<br><b>To: </b>"</span><a href="mailto:users@lists.opensips.org"><span lang=EN-US>users@lists.opensips.org</span></a><span lang=EN-US style='color:black'>" <</span><a href="mailto:users@lists.opensips.org"><span lang=EN-US>users@lists.opensips.org</span></a><span lang=EN-US style='color:black'>><br><b>Subject: </b>[OpenSIPS-Users] register phone in same local network as opensips</span><span lang=EN-US style='font-size:12.0pt;color:black'><o:p></o:p></span></p></div><div><p class=MsoNormal><span lang=EN-US style='font-family:"Times New Roman",serif'><o:p> </o:p></span></p></div><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Hello,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>I have opensips 2.3 beta, along with a local phone both running inside and behind the same router.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>I have one port forwarded for opensips to listen on and a port range forwarded for the local phone.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>There is a remote phone in another domain behind another router, also with a port range forwarded.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>I am using stun for both phones and this resolves the correct IP domains for each phone.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>With stun implemented and saying it is full cone on both phones. The phones can now ring each other.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>The local phone can call the remote phone and there is two way audio.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>When the remote phone calls the local phone, there is neither way audio.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Invites from the remote phone always appear with the correct expected provisioned sip and rtp ports.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>I would expect that the local router is changing the local phones sip contact port when it registers.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>When I look at a sipgrep capture of an outgoing invite both the sip and the rtp ports are changed.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>I am not all that sure where in the process or even why the rtp port for the invite has been changed.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Inbound calls then of course end up sending and returning rtp through non forwarded port ranges.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>What I would like to understand is how to make an inclusion, when any local phones register,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>that will allow the outgoing contact details to show the phones actual provisioned sip ports.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>With that correct, in the outgoing invite, the rtp streams would then normally be within the<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>expected range of ports opened and forwarded to the phones and that would solve the audio.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>I am looking for working examples, but I have not turned up enough specific information about just this.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Knowing better where and how to start and the names of what I am looking for would be most helpful.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Thankyou<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Alex<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p></div></body></html>