Common VoIP poor call quality symptoms and causes

We get asked this a fair bit here, I’ve compiled a quick list of the most common symptoms and causes with some possible solutions too:

Symptom: caller or callee hearing any of the following – clicks, periods of silence (voice stopping and starting), “robotic” sounding voice. This is by far the most common issue.

Cause: packet loss, can be due to lots of things, insufficient Internet bandwidth, lack of QoS on a connection shared with data, faulty network equipment (can include poor cabling), problems at the ISP.

More on bandwidth: a normal g711a VoIP call will require approximately 100kbps in both directions on the wire. The actual audio part is 64kbps but then you have to factor in RTP headers, IP headers, UDP headers etc… So it doesn’t matter if you have a 10Mbps Internet connection if this only has 256kbps, then you will only ever get two VoIP calls and even this assumes you are doing pretty much nothing else with it.

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Symptom: crackling during phone calls.

Cause: This is going to be a hardware issue with something like the phone’s handset or handset cable. Or headset if a headset is being used.

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Symptom: echo during calls, either the caller or callee hears their own voice coming back at them a fraction of a second after they spoke.

Cause: the fault usually lies with the person not hearing the echo. I.E. if a person you have phoned complains of echo then it is more than likely something on your phone causing it. The most common cause is people having handset volume turned up miles too loud, microphone gain too high or using a very poor quality handset or headset. It’s normally going to be an acoustic problem. Although it can also be caused by phones with extremely poor quality hardware and not very good echo cancellation routines (was common in the very early days of VoIP).

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Symptom: calls are too quiet, people who call me can’t hear me or I can’t hear them. My phone handset/headset volume is at full.

Cause: this is usually caused at the point where a call is translated from one format to another, such as inside a PBX converting an ISDN call to a SIP/RTP call. The fault needs to be fixed where at the cause rather than trying to mask the problem by turning handset volume up too loud (as this is likely lead to other problems such as echo on other calls). Most PBX systems will have settings to adjust gain levels when converting calls from one format to another. If this only occurs when using a headset, then check you are using the correct one for your phone.

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Symptom: when using a headset, the person I am talking to can hear a buzzing on the line. For those in the know, it is a 50Hz “mains buzz”.

Cause: this is caused by electrical interference being picked up by your headset’s cable. Causes can be faulty electrical equipment (computer, computer screen etc…) nearby. One solution is to ground your phone somehow. Either install fully shielded network cabling (which isn’t much use if you already have unshielded UTP cable installed throughout your building!). Or power your phone with a fully earth power supply, these are identifiable as they will have a 3-pin “IEC” connector from the wall socket to the power brick. Fortunately there is an easier answer which is to buy a headset which has a better quality shielded microphone.

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White-label parcel tracking

ProVu have offered a white-label order fulfillment service for many years.

As a reseller, your customer orders from your site, you order from us and we ship as if it had come from you. “Simples”.

What happens when they want to know the delivery status of their order? Using our white-labeled parcel tracking service you can offer your customer a link to a page with real-time tracking.

If you are interested in this service, please contact our sales team for pricing information.

ipv6

IPv6 Certification Badge for timprovu

IPv6 is becoming a big topic at the moment.

We expect IPv4 addresses to run out this year. This isn’t so bad as we expect ISPs will start selling addresses to each other which will even out supplies for a few years.

But the future is IPv6. At ProVu we have had our network IPv6 activated for about 5 years. This has given us lots of experience.

We are also pushing all of our suppliers to update us on their IPv6 status. Ideally, all phones would support IPv4 and IPv6 dual stacked at same time.

We’ll update you as time goes on.

In the meantime, you could test your IPv6 connectivity using http://test-ipv6.com/

Disable SIP ALGs on Draytek Routers

Many commercial routers now a days are coming with SIP ALG`s turned on by default. This feature is suppose to help with Nat related issues but in majority it break the implementation and cause issues such as one way audio, lack of incoming calls, registration issues and etc.

Recently, We have received couple of calls regarding IP phones not working behind DreyTek routers. The reason is same SIP ALG.

If you are a victim of this feature please disable it using the following procedure.

Telnet into the router and enter the following command.

1. > sys sip_alg 0 — Disables sip alg

2. > sys commit — Apply changes

3. > sys reboot — Reboot router

Once router is back online, reboot the ip phone or press re-register.

PROVU DISTRIBUTES AASTRA SIP PHONES

aastra
aastra office

We are delighted to announce today that ProVu have been appointed as Distributor for the Aastra range of SIP phones and accessories.

Aastra is recognised globally as a leading supplier of enterprise telephony equipment and their SIP handset range is a natural fit that enhances the ProVu product portfolio. Their excellent SIP handset range is both stylish, functional and competitively priced.

With ProVu entering a phase of rapid growth and seeing demand for its automated fulfilment and provisioning services increase, we feel it is an essential part of our strategy to be able to offer our customers these services across an even wider range of quality products.

Stock and Pricing
All the Aastra models are in stock and available for next day delivery, you can view our trade pricing at our on-line trade price list below.

http://pricing.provu.co.uk/

(If you do not have a log in for the price list just email us and let us know.)

Ordering
As with all our products Aastra phones can be ordered by, phone or email.

Or for all our customers using our ProSys on-line ordering the Aastra phones are now listed and can be ordered via the ProSys on-line ordering system.

Provisioning
Our automated Provisioning platform has now been developed to accommodate the Aastra phones.

So for all our service provider customers and PBX installers wanting to deploy phones in volume or on a regular basis you can now order Aastra phones and have them provision automatically out of the box.

For more Information on the range of Aastra phones please visit:

http://www.provu.co.uk/aastra.html

Thomson Routers

ProVu are now stocking the Thomson TG585 V8 routers.

This is primarily for ITSP and installer customers so they can buy the phones and router from one supplier.

We can also set the ADSL username and password in the router before shipping.

Protalk SIP Door Entry Phone with Gradwell

The scenario is that you are using one of our ProTalk Door Entry SIP phones with a Gradwell account and forwarding the call to a mobile phone. You might want to do this to talk to people at your door when really, you aren’t in or for when you aren’t near a desk phone in your place of work.

If you still want to operate the relays in the door phone (either to activate a door opener, turn on a light etc…) then you type in a certain sequence on the phone you are using. This sequence is transmitted back to the door phone using what is known as DTMF tones. In order for this to work with Gradwell accounts we’ve found that you need to use their Outbound Proxy.

Firstly, make sure you are on a recent firmware (v1.48 at the time of writing), you can get this from ProVu if you need to upgrade. Then set the device up as follows in the SIP Parameters page:

  • SIP Proxy Server Address – nat.gradwell.net
  • SIP Proxy Server Port – 5082
  • SIP Registrar Server Address – sip.gradwell.net
  • SIP Registrar Server Port – 5060

Then your username/password as Gradwell will have given you.

This may also be the case for other providers as well as Gradwell, so if you are getting one-way-audio problems or DTMF problems in general, ask them about Outbound Proxy.

More info on the ProTalk SIP Door Entry range here.

Differences between SARK and SnomONE PBX

I’m being asked which is better, SARK or SnomONE a lot at the moment. Rather than a Harry Hill style fight, I’d thought up a quick comparison list. I don’t believe either system is “better” than the other, they have their own advantages which work in certain scenarios. Horses for courses…

Advantages of SnomONE over SARK:

– 10 user version free and downloadable.

– Windows, Linux & Mac versions (nice easy installers for each although I’m yet to try the Mac version as I don’t have a Mac).

– software only (some people see this as an advantage)

– PnP snom phone config (even slicker than Adopt in SARK)

Advantages of SARK over SnomONE:

– complete solution inc hardware (some people see this as an advantage)

– supports any SIP/IAX etc… end point (snomone is basically snom phones only apart from the odd one or two periphery devices)

– extensible : you can write your own Asterisk code and modify existing code.

– 100s if not 1000s of 3rd party bits of software support Asterisk to do all sorts of things. Including call centre reporting, operator panels, call billing etc…etc…

– choice of telephony interfaces for connecting to ISDN lines.

– custom code gives options for much more complicated call routing scenarios.

That’s all I can think of right now, this isn’t an exhaustive list 🙂

Snom joins compatible devices program for Microsoft Lync 2010

Following on from the recent news that the Snom300 is compatible with OCS 2007R2, Snom have also joined the “Compatible Devices Program for Lync 2010”. This means that it is continuing it’s development of “OCS” firmware to include the new features that will be present in the future Lync release.

Lync is the new name Microsoft has given to the next version of OCS, it was generally referred to as Wave14 until this official name was released. Lync is currently at the public beta stage so not yet ready for production use.

ProVu will be testing Lync with Snom phones in due course and Snom themselves will be developing the new features possible with Lync.