What I mean is you&#39;ll see two separate m=audio lines within the callee&#39;s SDP, one for &#39;RTP/AVP&#39; and one for &#39;SRTP/AVP&#39;. If there is no m=audio line for RTP/AVP the caller won&#39;t know of a port that&#39;s expecting RTP. So if the callee only sends SRTP/AVP the caller can&#39;t send RTP.<div>


<br></div><div>Can you show us the SDP being sent by the phone?</div><div><br></div><div>-Steve</div><div><br></div><div><br></div><div><br><div class="gmail_quote">On 7 February 2013 11:01, Levend Sayar <span dir="ltr">&lt;<a href="mailto:levend.sayar@karel.com.tr" target="_blank">levend.sayar@karel.com.tr</a>&gt;</span> wrote:<br>


<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">



<div dir="auto">
<div>Thanx Steven.</div>
<div><br>
</div>
<div>Caller makes the offer for SDP but callee chooses whatever it wants. So caller can offer SRTP but callee can prefer not to talk encrypted. In our case I want FS to choose non secure media.</div>
<div>Phone will offer SRTP on the conference call but FS must prefer RTP, not SRTP.<div><br>
<br>
<div><span>İyi çalışmalar.</span></div>
<div><span><br>
</span></div>
<div><span>_lvnd_</span></div>
<div> {^_^}</div>
<div><br>
</div>
<div>KAREL`UCAP</div>
<div><br>
</div>
</div></div><div><div>
<div><br>
On 7 Şub 2013, at 11:13, &quot;Steven Ayre&quot; &lt;<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type="cite">
<div>It&#39;s also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there&#39;d be nowhere to send insecure RTP.
<div><br>
</div>
<div>-Steve</div>
<div><br>
</div>
<div><br>
<br>
<div class="gmail_quote">On 6 February 2013 16:09, Levend Sayar <span dir="ltr">&lt;<a href="mailto:levend.sayar@karel.com.tr" target="_blank">levend.sayar@karel.com.tr</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF">
<div>Thanks Daniel for the reply.</div>
<div><br>
</div>
<div>I tried</div>
<div><br>
</div>
<div>&lt;action application=&quot;set&quot; data=&quot;sip_secure_media=false&quot; /&gt;</div>
<div><br>
</div>
<div>But did not work. Upon your reply I also tried</div>
<div><br>
</div>
<div><span>&lt;action application=&quot;set&quot; data=&quot;secure_media=false&quot; /&gt;</span></div>
<div><span><br>
</span></div>
<div><span>But did not work either. I am doing something wrong ?<br>
</span><br>
<div><span>İyi çalışmalar.</span></div>
<div><span><br>
</span></div>
<div><span>_lvnd_</span></div>
<div> {^_^}</div>
<div><br>
</div>
<div>KAREL`UCAP</div>
<div><br>
</div>
</div>
<div>
<div>
<div><br>
On 6 Şub 2013, at 18:00, &quot;Daniel Ivanov&quot; &lt;<a href="mailto:sertys@gmail.com" target="_blank">sertys@gmail.com</a>&gt; wrote:<br>
<br>
</div>
<div></div>
<blockquote type="cite">
<div>
<p>Of course you can. Just set the secure_media var to false and you will be srtp-free in sip.</p>
<div class="gmail_quote">On Feb 5, 2013 6:06 PM, &quot;Levend Sayar&quot; &lt;<a href="mailto:levend.sayar@karel.com.tr" target="_blank">levend.sayar@karel.com.tr</a>&gt; wrote:<br type="attribution">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<u></u>
<div>Hi all.<br>
<br>
I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can<br>
join a conference. FS can talk to each peer with SRTP or not depending on the phone itself.<br>
<br>
My question:<br>
<br>
Is it possible to disable SRTP on FS ?<br>
<br>
I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not.<br>
<br>
TIA<br>
<br>
<br>
<table cellspacing="0" cellpadding="0" width="100%">
<tbody>
<tr>
<td>
<pre>-- 

_lvnd_
 {^_^}

</pre>
<br>
<br>
</td>
</tr>
</tbody>
</table>
</div>
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