openh323 was segfaulting the box for me every couple of thousands of calls. So it just unstable :) On 29 June 2006 20:01, Mr.Surender Reddy wrote: > Dear Anton, > Why not using openH323 instead of IAX > can u please explain me this if u dont mind.if > possible.And can u give me ur best recomendation plz so > that may be i can implement to check that if that slove > the problem to me. > > regards > surender > > On 6/29/06, Anton <anton.vazir@xxxxxxxxx> wrote: > > Dear Anders, > > > > If you could have a look into the topic - that would > > help us very much. > > > > For those, having the audio-lost problem I would > > suggest the following way - which helps me, and brought > > audio lost somewhat down (though, it does not > > elliminate it completely, but you can offer a service > > atleast) > > > > ------------- > > I'm managed to make it > > not-so-extensive, by interconnecting two asterisk boxes > > via IAX2 with JitterBuffer enabled. While using it > > directly (chan_ss7) as gateway for SIP voip - the audio > > lost makes things commercially unusable completely. > > > > so my current scheme, which brings down audio-lost > > problem TELCO <-> [chan_ss7->IAX2 ] <-> [IAX2->SIP] <-> > > WORLD > > > > direct scheme > > TELCO <-> [chan_ss7 -> SIP ] <-> WORLD > > is commercially unusable > > > > that's why I'm sure making the JB or incorporating > > chan_zap's JB for chan_ss7 would solve the problem, > > since I have not heard the asterisk's PRI has that > > problem - but in meaning of communication with TDM > > cards - there is no difference. So the implementation > > is the reason. Maybe PRI/chan_zap guys know something > > chan_ss7 guys does not know > > > > I would add that current problem MOST LIKELY happens on > > satellite links mostly - when jitter may vary in > > 50-100ms > > > > What I also noted, while playing with chan_oh323 - > > which uses OPENH323 code, with self JitterBuffer, audio > > lost is MUCH lower, in comparision with SIP or > > chan_h323 (which DOES NOT HAVE JitterBuffer) so the > > packet arrival instability clearly affecting that. > > > > Anton. > > > > On 29 June 2006 17:49, Mr.Surender Reddy wrote: > > > Yes sir We tried Dell ,HP Machines but it endup with > > > the same audio lost . > > > > > > regards > > > surender > > > > > > On 6/29/06, Patrick <asterisk-list@xxxxxxxxxxxxxxxxx> > > > > wrote: > > > > On Thu, 2006-06-29 at 17:21 +0530, Mr.Surender > > > > Reddy > > > > wrote: > > > > > Dear sir, > > > > > We have tired all the options > > > > > avaliable on the net and the things like patches > > > > > and what ever the other users suggested but we > > > > > are unable to move this problem out .The voice > > > > > comes excellent but when the Choppy voice doesnt > > > > > continues always but may be for 10 to 30 sec for > > > > > 1 or 2 or 5 or 10 minutes it regurally comes and > > > > > goes.IF this problem is solved i can say that > > > > > chanss7 is the best as the voice we have seen no > > > > > other providers here could gives this quality we > > > > > are only loosing the market as we have this audio > > > > > lost problem and people doesnt want to buy the > > > > > callingcards at all.If any one could find a > > > > > solution that will be a great help for this > > > > > chanss7. > > > > > > > > Have you tried another server? > > > > > > > > Regards, > > > > Patrick > > > > > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by Easynews.com > > > > -- > > > > > > > > asterisk-ss7 mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-s > > > >s7 > > > > _______________________________________________ > > --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 -- With Best Regards, Anton -------- Anton V. Gnitko Technical Director Eastera Co. Ltd. +992 372 270101 +992 372 213627 http://www.eastera.net
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