Hi guys, I've done tons of experiments with chan_ss7 and my impression of having audio lost problem is that there is some mis-clocking issue. When there are calls over g711 codecs from the SS7 PC - I do not have Audio Lost trouble. And mostly that happens on the long latency links (satellite 600ms+) so right now I'm trying to get it working woth external codec translator via SIP->H323 with parallel translation form G711->g729 - so that's works. I guess that sifira guys also do not use the 723/729 codecs on the SS7 PC - so they did not experienced the trouble. So the partial solution - is to use a distributed system with SS7-VOIP on different PC's
I've tried to replace SIFIRA ss7_write code with the code similar to used in chan_zap -> with almost the same result (although with a little better behaviour, i think binded with the fact that in chan_zap the code does not attempt to write more than 160 bytes at once to the ZAP in the cycle) Any similar experience? _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7
