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/20080817/1dd9d741/attachment.htm>
