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. best regards tiago On Sun, Aug 17, 2008 at 7:02 AM, Manish Rana <manish.rana at gmail.com> wrote: > hey generally it depends on the network dealys............ > but i think generally 1000ms is an Ok ok but RTCP will genreally let u know > the jitter value > > On Sat, Aug 16, 2008 at 7:23 PM, Tiago Katcipis <katcipis at inf.ufsc.br>wrote: > >> and how bigger would have to be the jitter buffer? i cant use on the lib a >> too big jitter. >> >> thanks everyone for the help >> >> >> On Sat, Aug 16, 2008 at 5:25 AM, Manish Rana <manish.rana at gmail.com>wrote: >> >>> hey but i setting the jitter-buffer latency we can take care of the round >>> trip delays >>> and alsa will get mostly sequenced audio packets.... >>> >>> >>> >>> On Sat, Aug 16, 2008 at 9:04 AM, gulshan karmani < >>> gulshan.karmani at gmail.com> wrote: >>> >>>> hi all, >>>> Only issue cud be voice rendering which has constraints of round trip >>>> delays due to naec and in that case use of alsa plugin cud be a >>>> problem. >>>> Rgds, >>>> Gulshan >>>> >>>> On 8/15/08, Manish Rana <manish.rana at gmail.com> wrote: >>>> > Hi, >>>> > >>>> > 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........... >>>> > >>>> > BR >>>> > Manish >>>> > 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 C?sar Katcipis >>>> >> >>>> >> >>>> ------------------------------------------------------------------------- >>>> >> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>> >> challenge >>>> >> Build the coolest Linux based applications with Moblin SDK & win >>>> great >>>> >> prizes >>>> >> Grand prize is a trip for two to an Open Source event anywhere in the >>>> >> world >>>> >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>> >> _______________________________________________ >>>> >> Gstreamer-embedded mailing list >>>> >> Gstreamer-embedded at lists.sourceforge.net >>>> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded >>>> >> >>>> >> >>>> > >>>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.freedesktop.org/archives/gstreamer-embedded/attachments/20080817/b275ab82/attachment.htm>
