xfone 018 phone service and Linux

[e2xbegqsdyt21hfc]

I don't know the technical details or 018.
I do know that 012 provides you with a sip adapter that is plugged between  the bezeq/hot modem and your PC/ordinary phone. It could be that that adapter reserve some BW for the sip packets. In any case, my experience is that both 750K ADSl/cables is sufficient for a good quality phone service, with a lightly usage for internet and phone. The sip adapter I initially had was made in the US. I assume they use a US proven technology. My current sip adapter is made by AudioCodes. 012 claims that the sip adapter they provide should work with any ISP. I had two ISPs, and in my case their claim is justified.
There is also a rumor that bezeq's new generation network uses sip over copper for the phone services.

--- On Sat, 3/7/09, Arie Skliarouk <skliarie at gmail.com> wrote:

> From: Arie Skliarouk <skliarie at gmail.com>
> Subject: Re: xfone 018 phone service and Linux
> To: "linux-il" <linux-il at cs.huji.ac.il>
> Date: Saturday, March 7, 2009, 11:21 PM
> Hi,
> 
> On Thu, Mar 5, 2009 at 13:54, Gilad Ben-Yossef
> <gilad at codefidence.com>wrote:
> 
> >  Arie Skliarouk wrote:
> >
> >
> >
> > How do they solve the latency problems inherent to any
> internet connection?
> >
> > The round trip time to a well connected (read: 0%
> packet loss) server in
> > Israel from an Israeli ISP, where Israel here is
> defined as "connected to
> > the IIX", is under 50ms.
> >
> 
> On a 150kbit upload ADSL upload of a 1500 bytes packet
> takes about 85ms. To
> send out the VoIP packet would take another 85ms (on
> condition that you have
> a really good QoS). This causes latency of 85-170ms with
> jitter 85ms. The
> VoIP speech (SIP protocol) has a lot of skips and is
> unacceptable.
> 
> Theory aside, I would like to hear first-hand experiences
> of people with 018
> before I commit for a year of phone service with them.
> 
> -- 
> Arie
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