>> 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 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
