[asterisk-dev] Jitter Buffer
Martin Vít
vit at lam.cz
Tue Oct 24 15:10:42 MST 2006
Slav Klenov wrote:
> Pavel Jerek, Martin Vit: The jitterbuffer should work for all cases,
> however this doesn't stands for the IAX channel (from and to). The IAX
> channel has its own jb (newjb by stevek). You can disable it setting
> jitterbuffer=no in iax.conf, but in this case you will have the very
> old iax2 jitterbuffer, which is hardcoded in chan_iax2.c and cannot be
> disabled by configuration. Thats why I (and russell, when ported it to
> trunk) omitted to include jb code in chan_iax2.c - you just can *not*
> have jitterbuffer when doing [some chan <-> Asterisk <-> IAX chan] in
> both directions! nor any generic jb (jbenable, jbforce, ...)
> configuration properties are read from iax.conf - IAX cannot enable
> the use of generic jb for its end and IAX cannot provide generic
> timestamp information (in struct ast_frame) to be used for dejittering
> on the other channel's end. This
> http://bugs.digium.com/view.php?id=8189 bug report seems very strange
> to me - it is not supposed the generic jb to work in this case at all
> (no matter what settings you have).
I would clarify this bug (8189).
Consider this scenario (ilbc)SIP -> * -> ZAP. I've traced how
translators are prepared for this: (ilbc)SIP -> ilbctoslin -> jitter
buffer -> asterisk bridge -> write Zap (translation is done BEFORE
dejjitering). The problem is when several frames come from SIP RTP very
fast (because of delay followed by burst packets). If i understnad
asterisk translator structures, there is one outbuffer used within this
translator where translated frames are stored. but it is not clear to me
if there is possible race condition and frames could overwrite this
buffer or if incoming RTP packets are passed or not passed concurently
to translator.
If I change translator path in ast_bridge to this: (ilbc)SIP -> jitter
buffer -> asterisk bridge -> ilbctoslin write Zap everything is correct
(translation is done AFTER dejittering)
>
> Slav
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.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.vasesit.cz/
mobil: 605 267 610
More information about the asterisk-dev
mailing list