Voice breaking up or fading in and out: 


Symptom: When I make an outgoing call to a customer, the customer complains that they cannot hear me as my voice fades in and out.  However I can hear the customer perfectly.

The customer likens it to the VoIP users voice fading in and out for a very short time in calls at the beginning.  It seems like the user has not got their handset close enough to their ear. What would be the cause of this?


As the customers voice is clear and not breaking up or fading and this only affects the callers voice i.e. the VoIP user - this means that from our VoIP platform end to the public network the connection is fine so this is not an issue with our VoIP platform.


  • Please check that the VoIP user is holding the handset or using their headset properly?  Check that the microphone on the headset is in the correct position?  
  • Please also check that they are not using the loudspeaker function on the phone?  


Once the above have been eliminated and all is correct, then this issue is most certainly with your internet service provider (ISP) as it sounds like packet loss which is causing the call to breakup or fade in and out and so you need to get hold of your internet service provider to report the fault.



Symptom: If the problem is that you are trying to dial out and when you get through to a customer their voice keeps breaking up or fading in and out then please do the following:


1. Please make sure you have turned on call recording so that you can listen back to any problematic calls, please click here to find out how to play back your call recordings.


2. Are you using the loudspeaker function on the phone?  If so please stop using this and use the handset instead - does the problem still occur?     Please confirm.


3. Please do a speed test on your broadband line at the times when this has occurred so that we can look at the results of this test to monitor the packet loss or jitter which could be the cause of this issue.  


4. Please do a Traceroute command click here to see how and send us the results back


5. You can also do a ping to your VoIP IP address - please click here to see how to do this and then email over the results to our Support Desk.


6. Please confirm your ISP provider, router make and model.


7. Do you share your broadband line for both web and VoIP use?  If you do not have your own dedicated broadband connection for VoIP then you need to ensure that you use a router with a quality of service (QoS) setting to prioritise voice traffic over regular traffic as this is an absolute must.  Without QoS you will encounter poor quality telephone calls regularly.


8. Have you disabled the SIP ALG setting on your router?  



Symptom: caller or callee hearing any of the following - clicks, periods of silence (voice stopping and starting), "robotic" sounding voice.

Cause: Packet loss - Packet loss occurs when one or more packets of data travelling across a network fail to reach their intended destination. It is measured as a percentage. Call quality deteriorates when this value exceeds 5%.


Packet loss can be due to lots of things such as insufficient Internet bandwidth, lack of quality of service (QoS) on a connection shared with data, faulty network equipment (can include poor cabling) and problems at the Internet Service Provider (ISP).

The above can also be caused by Jitter. This is one of the most important factors examined during the VoIP speed test. In basic terms, jitter is the difference between when packets of data are expected to arrive and when they actually arrive. 

 

This often has little impact when you are browsing the web or downloading an e-mail, but for a real-time application like VoIP it makes a big difference. In the world of VoIP, timing is everything, and when the timing of packets are constantly being received at unexpected times, an unstable voice connection can result. 

 

Watch carefully for this during the VoIP speed test – the lower the score the better such as for instance 2ms (milliseconds) is better than 4ms.

 

A high level of jitter will cause severe degradation in call quality. If you saw a high level of jitter after running the VoIP speed test, you should be aware that your connection may have problems that could prevent it from properly running a VoIP service. 


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 does not 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.



Poor sound quality:


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


The fault usually lies with the person not hearing the echo.  i.e. if a person you have phoned complains of an echo on the line then it is more than likely something on your phone causing it. 


The most common cause is people having the handset volume turned up far too loud or their microphone turned up too high or using a very poor quality handset or headset as this is normally caused by an acoustic problem. 


Although it can also be caused by phones with extremely poor quality hardware and not very good echo cancellation routines (this was common in the very early days of VoIP).



Symptom: Problems with call quality there is crackling on the line when making outgoing calls


When you have an issue with your VoIP line where you hear crackling on the line this can be caused by a number of reasons: 


1. This could be an issue with your network.  We therefore recommend that you do a speed test on your broadband line that runs your VoIP service.  


We would expect to see a high Jitter reading as a result of an issue with your network.  We would then recommend that you contact your ISP to see if they have a problem on the line. 


2. Crackling on the line could also be down to a hardware issue with something like the phones handset or handset cable. 


3. This could also be caused by a faulty headset if a headset is being used. Try another headset to see if this resolves the issue.


4. If using a Soft phone please see if by installing another Soft phone client you have the same issue.  


5. You could try replacing your Ethernet cable to see if that resolves the issue.  We recommend you use a shielded Ethernet cable to connect your phone/base station to your router/network switch.  



Symptom: when using a headset, the person I am talking to can hear a buzzing on the line. 

This is caused by electrical interference being picked up by your headsets 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 is not 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.  


If you are experiencing issues with call quality we ask you to do the following:


1. Turn on call recording.  To find out how to do this please click here.


2. Make sure you listen back to any problematic calls - keep a note of date and time and number to play back recordings.  Please click here to find out how to play back call recordings.


3. Are you using the loudspeaker function?  Please stop using this function and use the handset instead - does the problem still occur?


4. If the call recording is OK with no sound issues that you had reported and you can hear the call recording clearly then this problem likely to be between you and your internet service provider (ISP).


5. Have you disabled the SIP ALG setting on your router? 


6Do you share your broadband line for both web and VoIP use?  


If you do not have your own dedicated broadband connection for VoIP then you need to ensure that you use a router with a quality of service (QoS) setting to prioritise voice traffic over regular traffic as this is an absolute must.  Without QoS you will encounter poor quality telephone calls regularly.


7. Please let us know your router make and model, provide this information by emailing the details to our Support Helpdesk. 


Dependent on the results of the above you will need to contact your Internet Service Provider to report this issue so they can investigate this for you.



Calls cutting off:


Symptom: Calls cutting off mid-way through a conversation:


This sounds like a network issue where the packet loss is dropping, please see some guidance below:


1. Packet loss is the biggest issue on call quality – we use a UDP packets to transmit the data between us and if a packet of data arrives out of sync then it is dropped – this is to ensure the real-time of the communication.


One simple way to monitor the line would be to do a ‘simple’ ping in the background to your VoIP IP address.  Please click here to follow instructions on how to run a Ping DOS command to your VoIP IP address.


Please send us the results of the above ping test as we can then look at the % of packet loss and maximum times – if there is any loss this will have an impact on call quality and likewise high ping times are likely to show an underlying issue too.  


Please email the results of the Ping DOS command to our Support Helpdesk. 


2. Calls cutting off can also be router related, therefore please also disable the SIP ALG on your router - there will be different instructions for doing this based on the type of router and model number. 


If you are not sure how to do this then type into Google search Disable SIP ALG and then enter the make and model of your router and it should bring up instructions specific to your router.


3. Do you share your broadband connection for both web and VoIP - if you do you need to ensure you use a router with a quality of service (QoS) setting to prioritise voice traffic over regular traffic is an absolute must.  Without QoS you will encounter poor quality telephone calls regularly.


4. If you disable the SIP ALG on your router and still experience issues then try using a different router all together to resolve the problem.


5. Please check the firewall setting on your router to check this is not blocking your port, please make sure that both ports 5060 and 5065 are open.


6. Alternatively try switching from port 5060 to port 5065 in your phone settings click here to see how to do this. 


7. If you are unable to disable the SIP ALG on your router please try setting up a STUN server in your phone set up enter stun.crmdomain.com into the Stun Server address field. 



Symptom: Outgoing calls cut out within the first 10 seconds but then I can re-dial and the call is unaffected why!?


Usually when we have seen this problem it is a network issue, specifically a SIP ALG on the router which is resulting in packet loss of data. 


What usually happens is, your SIP server will send an ACK (acknowledgement) packet back to the phone to say that everything is OK and the call is in progress.


It sounds like this packet is not getting back to the phone properly the first time, so the phone waits 10 seconds and then hangs up the call because it does not think it is working.


Please disable the SIP ALG on your router - there will be different instructions for doing this based on the type of router and model number.


If you are not sure how to do this then type into Google search Disable SIP ALG and then enter the make and model of your router and it should bring up instructions specific to your router.  You may need to contact your ISP for advice on this.

 

If you disable the SIP ALG on your router and still experience issues then try using a different router all together to resolve the problem.

 


Symptom: Outgoing calls manually dialled via the handset cut off within the first 10 seconds, then re-dialled are successful, this only affects numbers that I manually dial but calls made via Click2Call are unaffected, why?


Without a SIP trace it is hard to be 100% sure what is happening here as the call is being made via the handset and not over the SIP server. Usually when we have seen this problem it is a network issue, specifically a SIP ALG on the router.


What is SIP ALG? Please click here to find out. 


What usually happens is the SIP server will send an ACK (acknowledgement) packet back to the phone to say that everything is OK and the call is in progress. 


It sounds like this packet is not getting back to the phone properly the first time, so the phone waits 10 seconds and then hangs up the call because it does not think it is working.


The reason your Click2Call feature works over the CRM software is because it is not actually making an outbound call from the Gigaset phone.


The CRM Click2Call will make the SIP server send out two calls, one call to you (which is why your own phone will ring) and one call to the person you are trying to call. 


So you are actually receiving an inbound call rather than making an outbound call. The SIP server will then join both of these call legs together so that both sides (you and your contact) can talk to each other.


To resolve this issue you need to disable the SIP ALG on the router.  You may need to contact your ISP to find out how to do this. 


To discuss any of the above issues further please contact our Support Helpdesk. 
























Please contact our Support Desk asap to report this issue giving us details of the VoIP line you have dialled out from, call time and number dialled so that we can check with our VoIP providers to ensure that we do not have a problem on the line.