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-ss7
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