[asterisk-ss7] chan_ss7 1.0.10 jitter buffer

Klaus Darilion klaus.mailinglists at pernau.at
Thu Apr 3 09:15:49 CDT 2008

marek cervenka wrote:
>>> Hi!
>>> I do not understand the reason for having a jitter buffer in chan_ss7.
>>> The audio is received in on a TDM line. Thus there is no jitter.
>> I think the same reason as chan_zap has a jitter buffer. As far as I
>> know the other side of the conversation needs a jb.
> yes. if you terminate from SIP(outgoing call to PSTN) you need jb at 
> chan_ss7 side

Ok. This is clear. But shouldn't the jitter buffer be implemented in 
chan_sip? How should chan_ss7 know if the audio is coming from a channel 
technology which causes jitter or not?


> PSTN <---(chan_ss7 w/jb) Asterisk SS7 <----SIP---- SIP phone
> in reverse direction is jb in the phone
> PSTN --->(chan_ss7) Asterisk SS7 ----SIP----> (jb) SIP phone
> ---------------------------------------
> Marek Cervenka
> =======================================
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-ss7

More information about the asterisk-ss7 mailing list