[asterisk-dev] Jitter Buffer

[Slav Klenov]

I want to make some small corrections to my previous message:

Slav Klenov wrote:
> If you don't want jb on the ZAP end, you set jbenable=no in 
> zapata.conf - that would disable all dejittering on the ZAP end. If 
> you want to have jb on the ZAP end always, including the cases where 
> not needed (if the other chan doesn't create jitter), you set 
> jbforce=yes. The same is for the SIP channel - it has
Actually if you set jbforce=yes in zapata.conf, you'll have jb on the 
ZAP end only if the bridged channel can create jitter (because it can 
provide timestamp information). But it will be created anyway because 
jbenable=yes - the jbforce setting doesn't have any effect on the ZAP 
channel. It make sense to use jbforce=yes only for VOIP channel and it 
will take effect only when bridged to another VOIP channel.
> The generic Asterisk jb is created mainly for VOIP <-> PSTN cases. In 
> general, you won't need jb in VOIP <-> VOIP and PSTN <-> PSTN cases, 
> but you *can* if you configure asterisk *appropriately*.
You cannot have generic jb in the PSTN<->PSTN case. The reason is the 
same - nor of the channels provides timestamp information.