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New user - high latency and quality problems

jwhite
Grafter
Posts: 33
Registered: 31-07-2007

New user - high latency and quality problems

I have been setting up a new account today and experienced a few problems along the way.  As a new user of the service, I am not sure about the quality expected, so perhaps someone can advise me.
Like a user on another current thread, I also experienced high latency, but on speech - up to 4 seconds, which makes conversation difficult!  I heard this on my first call to the echo (160) service, then on a test call to my land line.  I was thinking - surely it's not meant to be like this?  I was using the SJphone client so swapped to X-Lite (latest build) and the latency reduced to sub-second.  Going back to SJphone, it seemed to be the VoIP outbound that was delayed, with inbound OK.  I just used the settings for both as advised in the setup guides (except for the proxy, which was different in my welcome email), with other settings left as default.
I also experienced poor speech quality at first on SJphone, like another post's "choppiness", along with maybe some front-end clipping.  I made G711 the default and these problems went away.
Also, the registration on client startup seems to take up to a minute - should that be quicker?
Can someone re-assure me these problems are abnormal, perhaps due to network changes or bugs in the SJphone clients, or just my errors in setup?  Otherwise, I don't think I'll be a regular user.
7 REPLIES
Community Veteran
Posts: 6,249
Thanks: 430
Fixes: 39
Registered: 30-07-2007

Re: New user - high latency and quality problems

Welcome to the voip forum.
Quote
I made G711 the default and these problems went away.

What was the default codec before you changed it ?
X-Lite's default codec is G.711 ( well at least it is on my Linux version )  which probably explains why the difference in the two.
TBH its a while since I used a Softphone in anger, as my main voip use is via a Siemens C475IP dual phone. In terms of speech quality on voip, what I can
say is that its difficult to tell the difference on my C475 between a landline call and a voip call.
As a matter of interest what sort of speeds do you get on your ADSL line ( router stats would be useful ) ?  since they could influence voip quality as could the router.
We've seen some issues recently with Netgear firmware.
Hope that helps ( a little )
puddy
Grafter
Posts: 1,571
Registered: 10-06-2007

Re: New user - high latency and quality problems

Sorry to here you are having problems with VoIP service
Like misterw I gave up on using a sofphone and brought VoIP hardware phone instead.  Using hardware phones I found that you cannot tell the difference between VoIP calls or BT landline calls.
VoIP hardware can start from has little as £15-£20 for a Linksys PAP2 Adaptor (eBay) or find a cheap speedtouch 716wl v5 £14 or Speedtouch 780wl £20-£30 (eBay).  I also own a Siemens dect ip phone  I like this  best it’s the only VoIP phone my partner can use.  They start from £50-£60 new. Or you may already have a Nokia Mobile Phone that can make VoIP calls. In addition Grandstream have a great range of VoIP phones too.
If you use the above VoIP products you don’t need to leave your computer on to make or receive VoIP calls but you need to have a spare port on the back of your router and need to leave your router on.
So from Free up to £60-£70  their plenty hardware to choose from
If you need any more help just ask!
Kind regards
Puddy
Ps When looking for Siemens VoIP dect phone’s make sure it as “ip” in its model number
Community Veteran
Posts: 1,139
Thanks: 4
Registered: 20-07-2007

Re: New user - high latency and quality problems

Hi jwhite
Like MisterW and puddy I'm now using a Siemens Gigaset phone and the speech quality is first class. I also occasionally use X-Lite, also without any problems. I did have problems with choppy sound on SJPhone which is why I switched to X-Lite.
When I was having problems with SJPhone, I was pointed to a site that allows you to test your line http://www.testyourvoip.com/. The results of my tests are documented here http://community.plus.net/forum/index.php/topic,74506.16.html.
Hope this helps - Bob
jwhite
Grafter
Posts: 33
Registered: 31-07-2007

Re: New user - high latency and quality problems

Thanks for the replies.
Mister W - SJphone started with the GSM codec at the top of the list, then iLBC below that.  I moved A-law to the top and speech tests were then OK.  I'm not sure which is the default on X-Lite, but I expect it was G711 - do you use the diagnostics to check?    My ADSL is normally pretty stable once it reaches its lower speeds - it had been running at 3776/448 for a few days.  The Router logs don't show any "events" at that period.
Puddy - I have set up a Linksys pair a while ago, but gave those away to a client.  It's probably the best solution, as it could connect with my landline as well.  What I'd really like, though, is a softphone that will link with TAPI into my contacts manager, so I can use that application to dial out as I can at present using a modem: however, a search didn't come up with anything that I could make work.  The PAP connected to the modem might let me select which line to use - e.g. *9 for VoIP.  PC independence isn't a big factor as I wanted to use VoIP only for outbound, work-related calls while at the computer.
Bob - I tried the test, thanks.  London showed a good connection, but Chelmsford failed.  I assume the Java applet has its own phone client. 
I am still puzzled by that outbound delay.  If a voice packet is delayed by much more than a second, shouldn't it be discarded?
Jim
itsme
Grafter
Posts: 5,924
Thanks: 1
Registered: 07-04-2007

Re: New user - high latency and quality problems

Quote from: jwhite
What I'd really like, though, is a softphone that will link with TAPI into my contacts manager, so I can use that application to dial out as I can at present using a modem: however, a search didn't come up with anything that I could make work.   

Are you aware that you can use the Directory in the portal (voip.plus.net) to setup calls?
Quote
This application will dial the Destination number exactly as if you had dialled it from Extension, with the exception that Extension will ring first and Destination will not be called until Extension is answered. If Reverse call order is selected, Destination will ring first.
The Destination may be any routable number, including another extension, or a UK or International PSTN number.

You can upload contacts to the portal, may require some messing around but it took me no more then 15 minutes for me to upload  350 contacts. May not be as clean as using TAPI as you have to maintain more then one address book.
Another advantage is for incoming calls the nickname in the address book is also transmitted with the CLI.
Community Veteran
Posts: 6,249
Thanks: 430
Fixes: 39
Registered: 30-07-2007

Re: New user - high latency and quality problems

Quote
I'm not sure which is the default on X-Lite, but I expect it was G711 - do you use the diagnostics to check? 

No , I just checked the order in the list.
Your ADSL speeds ought to be perfectly capable of supporting a decent voip connection but you might want to look at error counts just to see if they are excessive at any time, particularly late evenings.
Quote
What I'd really like, though, is a softphone that will link with TAPI into my contacts manager, so I can use that application to dial out as I can at present using a modem

Not quite what you were looking for but is this any use ?
http://www.arcosoft.com/vonalink/ScreenPopWin.htm
@itsme , Useful info on the Directory. I'd never realised about that.
jwhite
Grafter
Posts: 33
Registered: 31-07-2007

Re: New user - high latency and quality problems

Quote from: MisterW
Quote
What I'd really like, though, is a softphone that will link with TAPI into my contacts manager, so I can use that application to dial out as I can at present using a modem

Not quite what you were looking for but is this any use ?
http://www.arcosoft.com/vonalink/ScreenPopWin.htm

That looked as if it would do what I wanted.  Sadly, however, I couldn't get the TAPI bit to work (on Windows Vista) and there's no support.  Thanks, anyway.
Jim
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