<div dir="ltr">I mean, the phone may expect SRTP to be offered over the TLS socket.<div style>You were offering regular RTP profile "AVP" with no crypto in it and they rejected it as unsupported.</div><div style>
That makes me guess they require you to enable srtp "SAVP" + crypto tags or they will drop the call.</div><div style><br></div><div style>try adding {sip_secure_media=true} in the outbound call to the phone.</div>
<div style><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Feb 28, 2013 at 11:03 AM, Vittorio Guglielmo <span dir="ltr"><<a href="mailto:vittorio.guglielmo@gmail.com" target="_blank">vittorio.guglielmo@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Can you please explain to me what does it mean ?<br>
<br>
And, in case, how can I demonstrate it, cause I need to buy a lot of
clients to build a secured voip network.<br>
<br>
Thank you very much for your time and your effort on a so huge piece
of software ( I bought all of your books ).<br>
<br>
<br>
Vic<div><div class="h5"><br>
<br>
<br>
<br>
<div>On 28/02/13 17:42, Anthony Minessale
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">They probably do not accept calls over TLS that are
not SAVP</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Thu, Feb 28, 2013 at 6:36 AM,
Vittorio Guglielmo <span dir="ltr"><<a href="mailto:vittorio.guglielmo@gmail.com" target="_blank">vittorio.guglielmo@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> Anthony,<br>
<br>
sofia global siptrace on<br>
<br>
and in attachment are the two siptrace:<br>
<br>
<ol>
<li><b>101</b> ( Bria iOS 2.3.4 on Iphone 3GS) <b>===>
102</b> (Bria 3 release 3.5.0b on Mac OSX 10.7.5)</li>
<li><b>102</b> <b>===> 101</b></li>
</ol>
<p>Both of them have forced only ulaw (first) and alaw
(second), and without TSL/SRTP they call each other
without problems.<br>
</p>
<p><br>
The behavior is different in the two above cases:<br>
</p>
<p>When I call from 101 to 102 I get a <b>Originate
Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]</b><br>
due to, I believe a: <b>SIP/2.0 415 Unsupported Media
Type</b><br>
</p>
<p>When I instead call from 102 to 101 I get a <b>Originate
Failed. Cause: INCOMPATIBLE_DESTINATION</b><br>
with a : <b> SIP/2.0 488 Not Acceptable Here</b><br>
</p>
<p><br>
</p>
<p>Vic<br>
<br>
</p>
<div>On 28/02/13 00:23, Anthony Minessale wrote:<br>
</div>
<blockquote type="cite">
<div>sofia global siptrace on</div>
<div><br>
</div>
</blockquote>
<br>
</div>
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