<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<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">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 class="Wj3C7c"><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>
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