Still failling to achieve clear sound thru ADSL

Asterisk Asterisk Support
1

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.

2

Basicly you need to isolate the problem.

Two possible issues are are using a VPS with G729 codec,

and tunnelling the traffic through a VPN

or even a combination of both

Ian

3

Are u telling that g729 not work correctly with VPS ?

How can i isolate the issue ?

4

The best VPN is none. Second to that, the best VPN is one that encrypts packets, but maintains them as UDP. A VPN that tunnels through TCP will have problems prioritising the VoIP traffic. Prioritisation also requires help from your ISP.

The most likely problem is that you are sharing the ADSL link with other traffic and the VoIP is not getting priority, or is being delayed by TCP retransmissions, at the VPN level.

I don’t know what you mean by the sound going low. No transmission problem should affect amplitude.

5

Thanks for your repply david.

My frnd adviced me to use IAX2 with IAX encryption. Is it secure ? Cant that IAX2 encryption to detect ISP the voip traffic ?

Thx.

6

Please try re-phrasing this, or giving it in your mother tongue.

7

Please try re-phrasing this, or giving it in your mother tongue.[/quote]

Sorry david.

The thing is i need to hide my voip traffic from ISP. It should go via ISP as something else. Thats my main goal to use a VPN.

8

I think im having considerable packet loss. Hear i atatched it ? is thr anything i can do for this kind of situation ?

9

If skype call connect with crystal clear voice over ADSl link, but why we cant use same ADSL link to connect two asterisk server, atleast one channel at a time to get a good quality vocie quality ?

10

You also have a rather large round trip time.

Note, though, that ping is not a good test of packet loss, as ping packets tend to have a vary high priority for being discarded.

11

Hi david,

May be this test will usefull.

scribd.com/doc/37090887

Thx

12

Hi

You are testing two things here and then saying that as one is good so should the other,

Your skype tests and comments on teh quality are based on a direct connection to skype, you poor quality calls are via a vpn and are end to end. I assume you are in pakistan or the UAE or similar as not many other countrys block voip. and im supprised they are blocking port 4569 (iax)

if that 222ms is teh time to the othe rserver then you are on the limit and include into that encapsulting into the VPN you will have to accept the quaity issues you have.

if you are using iax then ias2 show netstats will give you stats on the calls

Ian