[gst-embedded] Gstreamer in VOIP

Manish Rana manish.rana at gmail.com
Fri Aug 15 10:16:50 PDT 2008


as far as the real time constrain is concerned, this should be taken care by
RTP and add the required delays shall be added by RTP, that is  gstrtpbin
plugin in gstreamer.
On addition to this gstreamer will give u flexiblity to create the pipelines
as your requirements. You can have minimal elements as well or add more to
get the better audio. (like audio resample and audioconvert can be optional)

And if i am not wrong Gstreamer is used successfully in maemo for the VoIP
application, and there is Farsight plugin available, which is optimised.

I am sorry if I have any wrong info... Please correct me....

Also please add more on the same...........

On Fri, Aug 15, 2008 at 3:58 PM, Tiago Katcipis <katcipis at inf.ufsc.br>wrote:

> I'm working in a project using voip on a software and in embedded systems,
> its more like a lib, that is used by a high level software for PC and in
> embedded systems. Actually everything is done in one single gigantic
> function, now we are working on creating a more readable and expandable lib,
> so we started to build the lib using pipes and filters patterns. That's when
> the idea of using gstreamer came, but since gstreamer is usually used on
> media players i would like to know if it is good to be used on real time
> voip systems that rely heavily on time to work properly. Is Gstreamer a good
> lib to build this type of application? If it is who would be the best place
> for me to start reading about it (using gstreamer on voip) ?
> sorry if i asked something stupid... I'm just starting on the job and don't
> have to much experience, sorry for the lousy English too :-)
> Best regards
> Tiago César Katcipis
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