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? regards klaus > > 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 _______________________________________________ --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
