<div dir="ltr">i will, if i find anything usefull for you guys of course...im just starting on voip...so im very newbie, it will be hard to me to find something interessting or new to you guys, but if i find i will let you know of course.<br> <br>im very thankfull for the help... if theres anything that i can do to help...ill do.<br><br>better start readind :-)<br><br>best regards<br><br>Tiago<br><br><div class="gmail_quote">On Sun, Aug 17, 2008 at 2:40 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><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html" target="_blank">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html</a></div> <div> </div> <div><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpsession.html" target="_blank">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpsession.html</a></div> <div> </div> <div>will help you </div> <div>also you can use the README file of gstrtpbin.</div> <div>Think that will be good to start...after that you can read RFC's </div> <div> </div> <div>All the best....Please share your findings as well as querries, that is how most of us learns :)</div> <div> </div> <div>Thanks a lot </div> <div>Manish<font color="#888888"><br><br></font></div><div><div></div><div class="Wj3C7c"> <div class="gmail_quote">On Sun, Aug 17, 2008 at 8:21 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">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><font color="#888888"><br>tiago</font> <div> <div></div> <div><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" 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"> <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> <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></div></div></blockquote> </div><br></div></div></div> </blockquote></div><br></div>