<div dir="ltr">hmm thanks for the help guys, my last question is where to start? i know that i can always try google, but maybe i will loose time reading the wrong stuff, if you point me on the right direction i would appreciate.<br> <br>best regards <br><br>tiago<br><br><div class="gmail_quote">On Sun, Aug 17, 2008 at 7:02 AM, 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>hey generally it depends on the network dealys............</div> <div>but i think generally 1000ms is an Ok ok but RTCP will genreally let u know the jitter value<br><br></div><div><div></div><div class="Wj3C7c"> <div class="gmail_quote">On Sat, Aug 16, 2008 at 7:23 PM, Tiago Katcipis <span dir="ltr"><<a href="mailto:katcipis@inf.ufsc.br" target="_blank">katcipis@inf.ufsc.br</a>></span> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;"> <div dir="ltr">and how bigger would have to be the jitter buffer? i cant use on the lib a too big jitter.<br><br>thanks everyone for the help <div> <div></div> <div><br><br> <div class="gmail_quote">On Sat, Aug 16, 2008 at 5:25 AM, Manish Rana <span dir="ltr"><<a href="mailto:manish.rana@gmail.com" target="_blank">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">hey but i setting the jitter-buffer latency we can take care of the round trip delays<br>and alsa will get mostly sequenced audio packets.... <div> <div></div> <div><br><br><br> <div class="gmail_quote">On Sat, Aug 16, 2008 at 9:04 AM, gulshan karmani <span dir="ltr"><<a href="mailto:gulshan.karmani@gmail.com" target="_blank">gulshan.karmani@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;">hi all,<br>Only issue cud be voice rendering which has constraints of round trip<br>delays due to naec and in that case use of alsa plugin cud be a<br> problem.<br>Rgds,<br><font color="#888888">Gulshan<br></font> <div> <div></div> <div><br>On 8/15/08, Manish Rana <<a href="mailto:manish.rana@gmail.com" target="_blank">manish.rana@gmail.com</a>> wrote:<br>> Hi,<br>><br>> as far as the real time constrain is concerned, this should be taken care by<br> > RTP and add the required delays shall be added by RTP, that is  gstrtpbin<br>> plugin in gstreamer.<br>> On addition to this gstreamer will give u flexiblity to create the pipelines<br>> as your requirements. You can have minimal elements as well or add more to<br> > get the better audio. (like audio resample and audioconvert can be optional)<br>><br>> And if i am not wrong Gstreamer is used successfully in maemo for the VoIP<br>> application, and there is Farsight plugin available, which is optimised.<br> ><br>> I am sorry if I have any wrong info... Please correct me....<br>><br>> Also please add more on the same...........<br>><br>> BR<br>> Manish<br>> On Fri, Aug 15, 2008 at 3:58 PM, Tiago Katcipis <<a href="mailto:katcipis@inf.ufsc.br" target="_blank">katcipis@inf.ufsc.br</a>>wrote:<br> ><br>>> I'm working in a project using voip on a software and in embedded systems,<br>>> its more like a lib, that is used by a high level software for PC and in<br>>> embedded systems. Actually everything is done in one single gigantic<br> >> function, now we are working on creating a more readable and expandable<br>>> lib,<br>>> so we started to build the lib using pipes and filters patterns. That's<br>>> when<br>>> the idea of using gstreamer came, but since gstreamer is usually used on<br> >> media players i would like to know if it is good to be used on real time<br>>> voip systems that rely heavily on time to work properly. Is Gstreamer a<br>>> good<br>>> lib to build this type of application? If it is who would be the best<br> >> place<br>>> 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<br>>> don't<br>>> have to much experience, sorry for the lousy English too :-)<br> >><br>>> Best regards<br>>><br>>> Tiago César Katcipis<br>>><br>>> -------------------------------------------------------------------------<br>>> This SF.Net email is sponsored by the Moblin Your Move Developer's<br> >> challenge<br>>> Build the coolest Linux based applications with Moblin SDK & win great<br>>> prizes<br>>> Grand prize is a trip for two to an Open Source event anywhere in the<br>>> world<br> >> <a href="http://moblin-contest.org/redirect.php?banner_id=100&url=/" target="_blank">http://moblin-contest.org/redirect.php?banner_id=100&url=/</a><br>>> _______________________________________________<br> >> Gstreamer-embedded mailing list<br>>> <a href="mailto:Gstreamer-embedded@lists.sourceforge.net" target="_blank">Gstreamer-embedded@lists.sourceforge.net</a><br>>> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded</a><br> >><br>>><br>><br></div></div></blockquote></div><br></div></div></div></blockquote></div><br></div></div></div></blockquote></div><br></div></div></div> </blockquote></div><br></div>