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<p>Hello,</p>
<p>it can play rtp files encoded in the format expected by the audio
codec. I used only when there was a re-invite (like putting the
call on hold).</p>
<p>Cheers,<br>
Daniel<br>
</p>
<br>
<div class="moz-cite-prefix">On 04.05.17 17:29, Yufei Tao wrote:<br>
</div>
<blockquote
cite="mid:CAJwP0iQ22oJeM7p=kuwhr7jTgzEk=piAnn+drhAbGzyWoREu5Q@mail.gmail.com"
type="cite">
<div dir="ltr">Hi,
<div><br>
</div>
<div>Just a quick question: if I use the Kamailio rtpproxy
module with the <span
style="color:rgb(0,0,0);font-family:helvetica,arial;font-size:12px;text-align:justify">Sippy
RTPproxy, can I stream audio within a call? Or is it only
possible before call audio is established?</span></div>
<div><span
style="color:rgb(0,0,0);font-family:helvetica,arial;font-size:12px;text-align:justify"><br>
</span></div>
<div><span
style="color:rgb(0,0,0);font-family:helvetica,arial;font-size:12px;text-align:justify">For
example can I mix/playback audio file into an existing
established call session at any time using functions
rtpproxy_stream2uac() and </span><span
style="color:rgb(0,0,0);font-family:helvetica,arial;font-size:12px;text-align:justify">rtpproxy_stream2uas()
somehow? If it is possible, does it need to be triggered by
an in-dialog message from either UAC or UAS?<br>
</span></div>
<div><span
style="color:rgb(0,0,0);font-family:helvetica,arial;font-size:12px;text-align:justify"><br>
</span></div>
<div><span
style="color:rgb(0,0,0);font-family:helvetica,arial;font-size:12px;text-align:justify">Cheers,</span></div>
<div><span
style="color:rgb(0,0,0);font-family:helvetica,arial;font-size:12px;text-align:justify">Yufei</span></div>
</div>
<br>
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</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
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