Could you please run a test on http://tools.voxygen.co.uk for me ? email me (privately) the results.
It should give you an indication of how many channels you can run reliably using SIP. The packet loss you are seeing is ~10% which might be usable, but it is on the high side, as is a round trip time of 250ms. Tim. On 7 Sep 2010, at 08:16, sampath jayashantha wrote: > > However i tested the adsl with a ping test, i guess im having some > considerable packet loss with USA servers. i think that because my ADSL > quality. > > Hear i have attached the image. Is thr anything i can do from my side to > improve voice quality ? > > > On Mon, Sep 6, 2010 at 1:52 PM, Tim Panton <[email protected]> wrote: > There are others who would disagree, but, my advice would be to remove the > VPN, and run an IAX trunk between your > home machine and the freepbx. You can enable IAX encryption if > confidentiality is an issue. > > If you possibly can, enable trunking on the IAX interface, this will nearly > halve the bandwidth you need between the 2 servers. > IAX trunking however needs a zaptel (aka dahdi) timing source which might not > be available on your VPS. > > A rough calculation shows you _should_ be ok with 8 channels of 729 even > without IAX trunking. > > > > On 6 Sep 2010, at 09:06, sampath jayashantha wrote: > > > Hi all, > > > > May be this question shouldn't ask hear. But I think you guys can give me a > > clue. Because im fighting with this from long time. > > > > My setup ; > > > > I have a asterisk server + freepbx running on VPS where hosted with > > burstnet. My partner send the call to this server thru sip trunk with g729 > > codec. Also i have another astersk now server located at my home. This home > > located server connected to internet via ADSL link {dynamic IP} ( download > > = 2Mbps & upload = 512 Kbps). I connected this two servers using a sip > > trunk. This sip trunk is connected via hamachi VPN. Home server have a > > 8port FXO card which is connected to PSTN network. > > > > My partners call come to vps server and then routed to home server and > > terminate to pstn network. > > > > Problem; > > My problem is i cant maintain a qualiti call from my vps server to homer > > server. Im using g729 licene codecs. Sometimes voice breaking. sometimes > > sound is low. sometimes no issue. still im trying to figure out what is the > > problem. > > > > Normally maximum number of call at a time is 3. > > > > I testest OPEN VPN and HAMACHI. What is the best VPN forVOIP ? > > > > Testes codecs ; G729, g711, gsm What is suit for ADSL ? > > > > Im doing testing testing and testing to achive good voice quality. Still im > > failed. > > > > Please tell me whats the best configuration to achive good sound quality > > over ADSL link. I saw ILBC is good for ADSL than G729. Is it true ? > > > > Im wondering skype works really fine when i use the same adsl link. But > > sometimes one sip calls also not giving me good quality. > > > > Please advice. > > > > -- > > .......................................................................................... > > > > There is always some one who know more Than us out there. > > > > Wê Lïvê †ð §hårê : Wê Lðvê †ð §hårê > > > > > > > > SAM > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisknow mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisknow > > Tim Panton - Web/VoIP consultant and implementor > www.westhawk.co.uk > > > > > > > -- > .......................................................................................... > > There is always some one who know more Than us out there. > > Wê Lïvê †ð §hårê : Wê Lðvê †ð §hårê > > > > SAM > <Packet loss.JPG> Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk
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