[asterisk-dev] IAX2 and jitterbuffer problems
Martin Vít
vit at lam.cz
Thu Oct 11 03:05:44 CDT 2007
Pavel Jezek wrote:
> Russell Bryant wrote:
>
>> Pavel Jezek wrote:
>>
>>
>>> - not working when bridge two channles with different jb implementation
>>> - eg. sip/h323/skinny & iax
>>>
>>>
>> I'm not exactly sure what you mean here. You generally don't want to use a
>> jitterbuffer in this situation. You only want to use it at the endpoint.
>>
>>
> I have excessive jittery connection (wifi/cdma), endpoints (eg. ci$co
> phones/gateways) can't cover this excessive jitter, I must dejitter on
> asterisk and send to endpoint with smaller jitter...
> I think dejittering is also needed before codec translation eg.:
> sipA--(jittery
> connection)--(iLBC)asterisk1(alaw)----iax----asterisk2---(alaw)sipB
> I think, to be efective plc codecs algorithm, I must dejitter _before_
> dojing codecs translation, so I must dejitter on asterisk1, not
> asterisk2 or even endpoints
it depends, on which channel translation is. For PLC it have to be in
outgoing channel (after dejjitering).
> ...
> but it's not possible, because bridging channels with different jb
> implementations (sip vs. iax) on asterisk1
> all this will be clearly solved, when jb will be efective in incomming
> direction on _incomming_ channel, as many people expected
>
If you want dejjitering on incoming channel, this channel must doing
recode to do correct PLC.
> I think, kind of currently designed jb in asterisk is confusing for many
> people....
>
>
>>
>>
>>> - jb is applied in incomming direction, but on _outgoing_ channel! :-\
>>>
>>>
>> I wrote a patch yesterday that allows this to work, even when connecting to an
>> Asterisk application such as Voicemail or MeetMe. See this post for more details:
>>
>> http://russellbryant.net/blog/?p=17
>>
>>
>>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
--
Martin Vít
LAM plus s.r.o.
http://www.lam.cz/
Tel.: 605 267 610
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20071011/eb9576e3/attachment.htm
More information about the asterisk-dev
mailing list