<div dir="ltr">Yes, but I would like leverage the plethora of plugins available for gstreamer (audio filters, video, etc...)</div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/8/6 Federico Zamperini <span dir="ltr"><<a href="mailto:fzamperini@tiscali.it" target="_blank">fzamperini@tiscali.it</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Take a look at pjsip (<a href="http://www.pjsip.org/" target="_blank">http://www.pjsip.org/</a>), it is specifically designed for internet telephony, so maybe the functionality you are looking for are already implemented.<br>

<br>
Il 06/08/2013 12:47, Sven Heyll ha scritto:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="im">
Hi<br>
<br>
thanks for the answer. interaudiosink/src sounds interesting. It is one<br>
approach I was thinking about implementing it by myself.<br>
Yes, I would like to have a steady rtp stream between two telephone<br>
calls with the option to start, pause and resume playbacks and<br>
recordings via DTMF. Technically it seems unexpected hard to do with<br>
gstreamer, though<br>
<br>
But isn't interaudiosink actually a hack? I haven't read the sources of<br>
interaudiosink/src yet, but shouldn't elments have the ability to be<br>
paused/seeked independent of the rest of the pipeline?<br>
<br>
wbr<br>
Sven<br>
<br>
<br></div>
2013/8/6 Tim-Philipp Müller <<a href="mailto:t.i.m@zen.co.uk" target="_blank">t.i.m@zen.co.uk</a> <mailto:<a href="mailto:t.i.m@zen.co.uk" target="_blank">t.i.m@zen.co.uk</a>>><div class="im"><br>
<br>
    On Mon, 2013-08-05 at 16:55 +0200, Sven Heyll wrote:<br>
<br>
    Hi,<br>
<br>
     > this is what I would like to do:<br>
     ><br>
     ><br>
     > RTP/UDP SRC ---> GstAdder ----> Some Audio Processing ---> RTP/UDP<br>
     > SINK<br>
     ><br>
     ><br>
     > Every now and then I'd like to dynamically add a playback:<br>
     ><br>
     ><br>
     > RTP/UDP SRC ---> GstAdder ----> Some Audio Processing ---> RTP/UDP<br>
     > SINK<br>
     >                    /\<br>
     >      urdidecode ---||<br>
     ><br>
     ><br>
     > My question: How would I "pause" and "resume" the uridecoder in the<br>
     > second figure.<br>
<br>
    Do you really want to pause and resume the uridecodebin part, or add it,<br>
    play, and then remove it, and then later add another one again?<br>
<br>
    You could do something with the inter elements (interaudiosink/src).<br>
<br>
    Cheers<br>
      -Tim<br>
<br>
<br>
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