<div dir="ltr">I'm very interested on using gstreamer because the flexibility and quality. Thanks for the help, now that i know that it is possible i need a hint on where to start... reading about gstrtpbin plugin?<br>
<br>Best regards<br><br>Tiago<br><br><div class="gmail_quote">On Fri, Aug 15, 2008 at 2:16 PM, Manish Rana <span dir="ltr"><<a href="mailto:manish.rana@gmail.com">manish.rana@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div dir="ltr"><div>Hi, </div>
<div> </div>
<div>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.</div>
<div>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)</div>
<div> </div>
<div>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.</div>
<div> </div>
<div>I am sorry if I have any wrong info... Please correct me....</div>
<div> </div>
<div>Also please add more on the same...........</div>
<div> </div>
<div>BR</div>
<div>Manish <br></div>
<div class="gmail_quote"><div><div></div><div class="Wj3C7c">On Fri, Aug 15, 2008 at 3:58 PM, Tiago Katcipis <span dir="ltr"><<a href="mailto:katcipis@inf.ufsc.br" target="_blank">katcipis@inf.ufsc.br</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c">
<div dir="ltr">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) ?<br>
<br>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 :-)<br><br>Best regards<br><br>Tiago César Katcipis<br></div><br></div></div>
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