<div>Dear Anton,</div>
<div> 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.
</div>
<div> </div>
<div>regards</div>
<div>surender<br><br> </div>
<div><span class="gmail_quote">On 6/29/06, <b class="gmail_sendername">Anton</b> <<a href="mailto:anton.vazir@gmail.com">anton.vazir@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Dear Anders,<br><br>If you could have a look into the topic - that would help us<br>very much.<br><br>For those, having the audio-lost problem I would suggest the
<br>following way - which helps me, and brought audio lost<br>somewhat down (though, it does not elliminate it<br>completely, but you can offer a service atleast)<br><br>-------------<br>I'm managed to make it<br>not-so-extensive, by interconnecting two asterisk boxes via
<br>IAX2 with JitterBuffer enabled. While using it directly<br>(chan_ss7) as gateway for SIP voip - the audio lost makes<br>things commercially unusable completely.<br><br>so my current scheme, which brings down audio-lost problem
<br>TELCO <-> [chan_ss7->IAX2 ] <-> [IAX2->SIP] <-> WORLD<br><br>direct scheme<br>TELCO <-> [chan_ss7 -> SIP ] <-> WORLD<br>is commercially unusable<br><br>that's why I'm sure making the JB or incorporating
<br>chan_zap's JB for chan_ss7 would solve the problem, since I<br>have not heard the asterisk's PRI has that problem - but in<br>meaning of communication with TDM cards - there is no<br>difference. So the implementation is the reason.
<br>Maybe PRI/chan_zap guys know something chan_ss7 guys does<br>not know<br><br>I would add that current problem MOST LIKELY happens on<br>satellite links mostly - when jitter may vary in 50-100ms<br><br>What I also noted, while playing with chan_oh323 - which
<br>uses OPENH323 code, with self JitterBuffer, audio lost is<br>MUCH lower, in comparision with SIP or chan_h323 (which<br>DOES NOT HAVE JitterBuffer) so the packet arrival<br>instability clearly affecting that.<br><br>Anton.
<br><br>On 29 June 2006 17:49, Mr.Surender Reddy wrote:<br>> Yes sir We tried Dell ,HP Machines but it endup with the<br>> same audio lost .<br>><br>> regards<br>> surender<br>><br>> On 6/29/06, Patrick <
<a href="mailto:asterisk-list@puzzled.xs4all.nl">asterisk-list@puzzled.xs4all.nl</a>><br>wrote:<br>> > On Thu, 2006-06-29 at 17:21 +0530, Mr.Surender Reddy<br>wrote:<br>> > > Dear sir,<br>> > > We have tired all the options avaliable
<br>> > > on the net and the things like patches and what ever<br>> > > the other users suggested but we are unable to move<br>> > > this problem out .The voice comes excellent but when<br>> > > the Choppy voice doesnt continues always but may be
<br>> > > for 10 to 30 sec for 1 or 2 or 5 or 10 minutes it<br>> > > regurally comes and goes.IF this problem is solved i<br>> > > can say that chanss7 is the best as the voice we have<br>> > > seen no other providers here could gives this quality
<br>> > > we are only loosing the market as we have this audio<br>> > > lost problem and people doesnt want to buy the<br>> > > callingcards at all.If any one could find a solution<br>> > > that will be a great help for this chanss7.
<br>> ><br>> > Have you tried another server?<br>> ><br>> > Regards,<br>> > Patrick<br>> ><br>> > _______________________________________________<br>> > --Bandwidth and Colocation provided by
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