reasons for making call twice before it connect

[yaman]

i wish to know what are the possible reasons as why sometimes you need to dial the same number a second time befrore the call is connected, meaning sometimes you dial the first time and there is one way voice problem so you hangup than try again and the call is fine on the second trial

anyone ave any idea what are the possible reasons?

is it ISP?, is it bandwidth? is it router? is it carrier? is it.....?

thanks

[mberlant]

yaman,

I think the answer is "all of the above." The VoIP services that you and I enjoy travel over the public internet. That makes them subject to all of the vagaries, congestions and troubles that tend to come up along your packets' route(s) of travel. It's all out of our control, and it often manifests itself in garbled speech or disconnections.

If you start out with marginally more bandwidth from your ISP than you need to support VoIP, or a VoIP service provider who is trying to squeeze too much service out of his equipment, you will be that much more susceptable to this interference. I am not saying that you are in this predicament, just that it's the very problem that kept VoIP-over-public-internet from flourishing until very recently.

I've been designing VoIP networks for more than 10 years, and it's only in the last three or so that the technology has been viable over the public internet.

[yaman]

Originally Posted by mberlant

yaman,

I think the answer is "all of the above." The VoIP services that you and I enjoy travel over the public internet. That makes them subject to all of the vagaries, congestions and troubles that tend to come up along your packets' route(s) of travel. It's all out of our control, and it often manifests itself in garbled speech or disconnections.

If you start out with marginally more bandwidth from your ISP than you need to support VoIP, or a VoIP service provider who is trying to squeeze too much service out of his equipment, you will be that much more susceptable to this interference. I am not saying that you are in this predicament, just that it's the very problem that kept VoIP-over-public-internet from flourishing until very recently.

I've been designing VoIP networks for more than 10 years, and it's only in the last three or so that the technology has been viable over the public internet.

mberland,

thanks for kind response, in case we need to narrow it to the end user with the thinking that the issue is at the end user side, can you please clarify for me these few reasons that might create the problem (i mean problem where the caller calls a number and he has one way audio, he hangs up than try again and the call go through nicely on the second time)

1- bandwidth: the end user claims he has 100 Kbps upload and he is using G711 and he is not using the internet while making a call and he even has QOS on his router.
NB: the end user has even increased his upload bandwidth to 200Kbps and still have the same issue

2- traceroute: the end user has made a traceroute to the voip provider sip server and there is no packet loss and it was 145 ms

can it be the ISP? or can it be the 100Kbps is on the limit? can it be that after doing an upload test that it is hsowing 110 Kbps but in reality it is never reaching 100Kbps?

3- router: the end user is using a router which he did open the necessary port 5060 and RTP 5004 to 5005 since the end user is using a grandstream device, or linksys where RTP port forwarded are 16384-16482 all for UDP

can it be that the router is causing this issue?

4- User agent: the end user is using differnet devices such grandstream and linksys and others.

i would appreciate is you could take these issues step by step to help narrowing the problem

thanks

[mberlant]

Except for the unnecessary forwarding of ports to support a client device, everything you just stated sounds perfectly fine. 100kbps should be plenty more than the 80kbps required to support G.711 (64k payload + 16k overhead).

traceroute, like ping, is not a good indicator of VoIP performance. These two utilities send very short packets, which generally are less susceptible to diversion and corruption than the larger payload packets you send during a VoIP conversation. A better indicator is to point your browser to www.testmyvoip.com and see what their evaluation of your VoIP friendliness is.