"pausing" parts of a pipeline

[Federico Zamperini]

Take a look at pjsip (http://www.pjsip.org/), it is specifically 
designed for internet telephony, so maybe the functionality you are 
looking for are already implemented.

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