For voip calls is the codec choice or the carrier more important?

[andav]

Hi

I'm having issues with my voip phone. I use X lite and the carrier https://didlogic.com/ for sip minutes

Most of my calls are fine - but often the call drops after like 15 minutes. I'm using the G.711 codec.

I have a 2mb broadband line which rarely goes down. When the calls drip my internet is still up, so I think the issue is either I should use a low bandwidth codec like G.729 or switch carrier for the sip minutes.

Anyone got any thoughts or experience? Would calls using a low bandwidth codec like G.729 be less likely to drop than the G.711 codec - all other things being equal?!

I'm not too fussed about the call quality - I just don't want the call to drop when I'm on the phone to a current or potential client!!

Thanks

Paul

 

[TurricanII]

Is that 2mbit upload too? Perhaps go to speedtest.net and post the numbers.
Can you knock the audio quality down to 4kbit mono in your softphone to reduce the bandwidth?
Are you downloading anything heavy when the call drops, orperhaps windows updates?
You might need to look at your Internet router, perhaps a firmware update for it. Have others had problems with VOIP through that model? Can QoS (Quality of Service) be enabled to prioritise VOIP?
I would also launch a continuous ping at your VOIP server name from your PC. What is the latency/response time? Is that VOIP provider in America?
I might also use winmtr (a free tool) and have it trace the route to the VOIP server, and look for slowness/packet loss.

 

[cjd]

Is that 2mbit upload too? Perhaps go to speedtest.net and post the numbers.
Can you knock the audio quality down to 4kbit mono in your softphone to reduce the bandwidth?
Are you downloading anything heavy when the call drops, orperhaps windows updates?
You might need to look at your Internet router, perhaps a firmware update for it. Have others had problems with VOIP through that model? Can QoS (Quality of Service) be enabled to prioritise VOIP?
I would also launch a continuous ping at your VOIP server name from your PC. What is the latency/response time? Is that VOIP provider in America?
I might also use winmtr (a free tool) and have it trace the route to the VOIP server, and look for slowness/packet loss.

More likely to be a router problem than your codec. But it's impossible to diagnose unless you're using our network. Check that SIP ALG is turned off in the router.

If that doesn't work, connect to us and if you still have a problem give us a call - 020 7043 5555.

 

[it-comms]

The X-lite Soft Phone (Free version) will not support G729.

is there an option within the softphone for keep alive - if so enable it and test.

 

[leemason]

In my experience the provider is much more important that the codec used in most circumstances. Whilst G.729 can help if the provider supports it dropped calls are just as likely to be a general connection (possibly firewall as CJD mentioned) or service provider issue. As it-comms said X-Lite being the free edition does not support G.729. I believe it does support GSM (assuming the provider supports this as well).

Didlogic are a very low cost provider. This does not necessarily mean low quality but they probably buy and sell call minutes and connection from whichever provider gives them the cheapest rates so quality can be variable. It might also be that the server you are connecting to is in the US which is never a good ideal for VoIP as distance increases the latency. With VoIP latency, jitter, and packet loss are just as important (if not more so) that actual line speed. On an ADSL line it is uplink speed that is important as well as that is much lower than downlink speed.

 

[andav]

Thanks for all your comments, you guys are smart!

I'm using Didlogic because they provide a Singapore server and I'm making calls from the Philippines to the USA.

If there was another carrier that provided a server close-ish to the Philippines where I am I'd use them but havent found another carrier that does.

Regarding the upload speed I ran a test and got 0.5mb up.

I pinged the voip server 50 times and had an average response of 82ms with no pack loss so I don't think its this.

I actually tried using the Eyebeam softphone today and g729 codec. I had one hour long conversation which didnt drop once - before often it would drop 1-2 times an hour. I don't know if this was coincidence or not.
However the client did say the call quality wasnt great - though at least it didnt drop

The router is the standard one the ISP here provided http://www.aztech.com/prod_adsl_dsl5001en.html. If I thought it was the issue I'd replace it but I don't know.

I don't know if ALG is turned on nor how to turn it off.

I don't think the issue is windows updates which I've disabled, and I'm not downloading anything else when the calls drop. I don't think its the firewall either - I'm just on a home adsl connection behind windows firewall on windows 7.

I'm going to test using this eyebeam softphone with G729 this week, fingers crossed it makes a difference.

Paul

 

[it-comms]

the 82ms ping will not be good with voice quality

 

[leemason]

the 82ms ping will not be good with voice quality

That's not true. I have customers who have longer ping times than that and their VoIP works fine. Most modern equipment has a jitter buffer which can handle longer latency times and 82ms is will within the bounds of most equipment to handle.

I have used G.729 quite extensively and to be quite honest a lot of the stuff you hear about reduced quality is exaggerated. I've always found the quality to be good enough for most applications. It can sound a bit flat as the dynamic range is not as wide but it's generally OK.

If you want better quality anyway you might want to look at a hardware device such as a Snom, Yealink, or Aastra handset. This will give you a better quality call even with G.729 compared with a soft phone.

 

[VoIP2Go]

I think your switch from X-Lite to Eyebeam will do the trick. I don't think server location is important as I have Customers in Australia going through servers in the US without issues. If the issue continues you might want to run some tests with an alternative provider so that you are able to make comparisons. Also, investigate Direct Media versus RTP Relay ;-)

 

[it-comms]

Really?

40 milliseconds is optimum for jitter delay. anymore and call quality may suffer.

That's not true. I have customers who have longer ping times than that and their VoIP works fine. Most modern equipment has a jitter buffer which can handle longer latency times and 82ms is will within the bounds of most equipment to handle.

I have used G.729 quite extensively and to be quite honest a lot of the stuff you hear about reduced quality is exaggerated. I've always found the quality to be good enough for most applications. It can sound a bit flat as the dynamic range is not as wide but it's generally OK.

If you want better quality anyway you might want to look at a hardware device such as a Snom, Yealink, or Aastra handset. This will give you a better quality call even with G.729 compared with a soft phone.

 

[andav]

I have used G.729 quite extensively and to be quite honest a lot of the stuff you hear about reduced quality is exaggerated. I've always found the quality to be good enough for most applications. It can sound a bit flat as the dynamic range is not as wide but it's generally OK.

If you want better quality anyway you might want to look at a hardware device such as a Snom, Yealink, or Aastra handset. This will give you a better quality call even with G.729 compared with a soft phone.

Thanks Lee - yes in my short time testing with G729 I'm with you. Sound wise its fine - if as you said sometimes a little flat.

Do you know if its possible to integrating click to call with a hardware handset? At the minute I'm clicking on numbers in an Excel spreadsheet which then automatically dials the number in Eyebeam. I don't know if this is possible using a hardware phone

 

[cjd]

SIP ALG is an option in most routers, you have to log into your router, find the relevant switch and turn it off. Some routers have this option removed, sadly.

Without knowing what your router is there's not much more to say.

I don't think it matters which codec you use but 729 isn't supported by most ITSPs - personally I'd switch to 711, you have ample bandwidth. If that turns out to be no better, try GSM.

 

[it-comms]

This depends on which phone system you are using. We use and supply the 3CX Phone System and click to dial is also supported on desk phones via some software installed on the PC.

Do you know if its possible to integrating click to call with a hardware handset? At the minute I'm clicking on numbers in an Excel spreadsheet which then automatically dials the number in Eyebeam. I don't know if this is possible using a hardware phone

 

[leemason]

Really?

40 milliseconds is optimum for jitter delay. anymore and call quality may suffer.

Yes, really.

 

[leemason]

Thanks Lee - yes in my short time testing with G729 I'm with you. Sound wise its fine - if as you said sometimes a little flat.

Do you know if its possible to integrating click to call with a hardware handset? At the minute I'm clicking on numbers in an Excel spreadsheet which then automatically dials the number in Eyebeam. I don't know if this is possible using a hardware phone

It depends on the handset. For example if you have have a phone from Yealink, Cisco, Polycom, Snom, or Grandstream then you can use this software http://www.theteletrigger.com/ to implement click to dial. This works with Outlook but not sure about Excel.