The speech quality of VoIP depends considerably on the characteristics of the individual underlying network. If you have a poor Internet connection or problems on your internal network, you are likely to see various problems with your voice traffic, which may show up as poor sound quality, clipping of voice or complete loss of audio.
There are many tests that can be undertaken to assess the quality of your internet/network connections.
We suggest these test are performed at various times during the day to get an overall picture of you internet connection.
Running the trace route application can determine if you are having problems connecting to VoIPon servers.
This will also show if you have any Latency issues in your connection (round trip times).
Running the bandwidth.com test your VoIP application will give you an indication of the Mean Opinion Score that your connection is experiencing.
The detailed report that this application produces will also enable you to track down where your network connection is failing. This application can be found at http://www.bandwidth.com/tools/voipTest/.
As this application does not involve any VoIPon servers it will help you to determine if the problem is a connection to VoIPon or a problem in general with your internet connection.
It is best that you run this application from the same connection that your VoIP device is using.
The following are some of the things you need to look for in your testing of your network connection.
In case of congestion at some point of the network packets may arrive out of order or simply with considerable, and/or varying delay (delay jitter). An efficient speech communication cannot be carried out if the transmission delay becomes too large. Hence packets arriving too late for timely playback may be discarded by the receiver (packet loss).
Similarly, if a router in the network is faced with too many packets during a traffic-burst period, it may have to drop packets (packet drop). Latency figures should be as low as possible with a maximum value of 150ms, anything outside this maximum figure will result in unusable VoIP communication.
Advanced configuration and troubleshooting
To reduce the amount of Data sent through the network, most IP-systems dispose of an optional mechanism to detect whether the current frame contains speech.
If it does, it is encoded and sent across the network. If it does not, a comfort noise is commonly inserted in the periods of silence at the receive side so that the channel is not entirely muted, and the user is assured that the call is still active.
Voice activity detection (VAD) and comfort-noise generation may both lead to perceivable speech quality degradation. For example, VAD may cause clipping and hence loss of information.
In addition, VAD interacts with the effect of packet loss: On one hand, with VAD the loss distribution and rate will be different because less data is sent across the network.
On the other hand, packet loss in periods of no speech in the connection without VAD may lead to artefacts on the background noise transmitted from the send side, which could be avoided with VAD. We suggest that if you are experiencing clipping that you turn off VAD on your VoIP device or software.
The choice of codec that you make can have a large effect on your VoIP communications. To reduce the bandwidth costs of voice transmission, a variety of different codecs are in use.
VoIPon support the following codecs on their Asterisk based VoIP network: G711a/law (VoIPon preferred codec) G711u/law G729a/b. In Digital PSTN networks the G711 codec is typically used. This is widely thought to be the best quality codec to use.
VoIPon recommends the G711 codec as it is used internally and also when voice traffic is passed to the VoIPon PSTN gateway. Using G711 reduces the amount of coding and decoding voice traffic encounters across multiple networks.
When selecting your codec you need to take into account the amount of bandwidth each codec uses. G711 codecs are the most expensive when it comes to bandwidth, using on average, approx 80kbits/sec for each leg of the voice journey (up and down each use a maximum of 80kbits/sec).
G729 codecs use, on average, approx 32kbits/sec, making this codec ideal if you are having bandwidth issues (in order to use this codec you must request it is enabled on your VoIPon account).
If you are using the Single Line account you can use any codec that the originating and destination devices can support.