[gst-embedded] Gstreamer in VOIP

[Tiago Katcipis]

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 César Katcipis
>>>> >>
>>>> >>
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>>>> >>
>>>> >
>>>>
>>>
>>>
>>
>
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