Voice Over IP in UK - sipgate lag

Author

sipgate lag

Joe Bryant

2006-07-25, 7:11 pm

Hi -

I'm new to VoIP (and to this group!) so forgive me if I'm going over
old ground.

I'm in the UK. I use sipgate to make calls to PSTN landlines within the
UK. I have a Handytone-286 which is plugged into my WGT624 Netgear
router, which is plugged into my 2 meg Telewest cable internet
connection.

I find that the sound quality is good, and the service generally has
nice features, but the lag in the audio is pretty bad, even when I'm
not using the internet connection for other things. It's bad enough
that it sometimes makes it hard to converse normally. It seems to me
that sometimes it's worse than others, but I couldn't swear to it, nor
can I discern a pattern.

My question: is this normal behaviour for VoIP in general? Or should I
be able to get a non laggy conversation going?

Further: If it's sub-normal, is there any way to determine whether the
problem lies with sipgate or my own set-up?

Thanks in advance for any advice.

Joe

stephen

2006-07-25, 7:11 pm

"Joe Bryant" <tenminjoe@yahoo.com> wrote in message
news:1153852499.916371.155120@i3g2000cwc.googlegroups.com...
> Hi -
>
> I'm new to VoIP (and to this group!) so forgive me if I'm going over
> old ground.
>
> I'm in the UK. I use sipgate to make calls to PSTN landlines within the
> UK. I have a Handytone-286 which is plugged into my WGT624 Netgear
> router, which is plugged into my 2 meg Telewest cable internet
> connection.
>
> I find that the sound quality is good, and the service generally has
> nice features, but the lag in the audio is pretty bad, even when I'm
> not using the internet connection for other things. It's bad enough
> that it sometimes makes it hard to converse normally. It seems to me
> that sometimes it's worse than others, but I couldn't swear to it, nor
> can I discern a pattern.
>
> My question: is this normal behaviour for VoIP in general? Or should I
> be able to get a non laggy conversation going?

there is some inherent lag in IP Telephony (ie all the codecs commonly used
need 20 mSec or more to encode - which is the entire delay for a classic UK
phone call on the PSTN).
there is a fair bit more in the jitter buffer to allow for variable delays
between talker and reciever
and the network itself tends to introduce more delay than the conventional
phone system, since entire packets go thru store and forward at each stage,
rather than individual sound samples.

There are various standards for how much delay is "too much" - a common
accepted limit for high quality is 150 mSec 1 way delay (or 300 mSec round
trip if you need to measure it from 1 end of the link).

Once the delay goes above a threshold you start to have problems with it -
often a person will start talking when the other hasnt finished and so on.

it is possible to build IPT systems which operate well within these limits -
i have worked on PBX replacements over WANs and although it takes some care
it is relatively easy, esp if the system only works over a small geographic
area (like the UK say).
>
> Further: If it's sub-normal, is there any way to determine whether the
> problem lies with sipgate or my own set-up?

that is harder - there could be something in the phone setup that adds
delay, but you need advice from someone who knows that model.

it could also be your ISP - see what the delay is when you "ping" some UK
based services - a broadband test utility can often give you some info on
this kind of stuff.
i have used "Dan Elwells broadband speed test". on an NTL cable in Mcr
earlier this gives 3 results in the 17.8 to 19.5 mSec range - this is low
enough delay from NTL isnt going to be a problem.

The other option i suggest is you try a different IPT service and see if it
gets better.

try to pick one that explicitly supports your phone and can advise on
settings.
>
> Thanks in advance for any advice.
>
> Joe
--
Regards

stephen_hope@xyzworld.com - replace xyz with ntl