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Delay/Network Latency is inherent in VoIP technology since the VoIP packets have to travel from the origination point to the destination through the end users own network and the public internet. Some of these delays are fixed by the laws of physics but the majority of the delay or latency relates to either the public network or the internal network .
VoIP Delay or Network Latency
The latency is the time it takes from the individual VoIP packet to travel from the speaker to the destination. Unlike applications such as e mail, latency can cause serious quality issues with VoIP. It is generally considered that the maximum acceptable one way latency for VoIP is 150ms with a maximum 2 way lag of 250ms. Network latency can be measured by a simple ping test. For most ADSL or cable connections a round trip latency of less than 100ms should be expected.
High latency can cause issues such as echo, lost packets and jitter and can make a VoIP service almost unusable. With modern ADSL or cable the main areas where latency is likely to occur are within the local area network, the lat mile of the ADSL network or the router. If the latency is intermittent the cause is often congestion either in the LAN or within the Internet Service Provider.
Overcoming congestion in the local Area Network can be done by implementing QoS and giving priority to the VoIP Packets, or having a separate ADSL connection for the VoIP service.
Continuous high latency may be due to a faulty router, or switch within the LAN, distance from the local telephone exchange,, poor quality ADSL or highly contender ADSL. For best VoIP service it is desirable to have low contention with other ADSL users, specific business VoIP services such as those offered by V4B are designed to provide the lowest latency levels possible.
One other possible solution to the issue of latency is to try different VoIP codecs. Codecs both convert and compress voice into data and data back to voice. There are a number of codecs that have different compression ratios and therefore have different bandwidth requirements. The main codecs used in VoIP tend to be either G729 or G711. G729 has a more compressed signal tan G711 meaning it requires less bandwidth. The actual bandwidth requirement depends on the number of frames per packer and can vary from 25 kbps to around 50 kbps. G711 tends to require around 75kbps. As with all choices, there is a compromise between selecting G729 and G711. G729 is a proprietary offering and therefore is not free, although most IP devices will include the option. If bandwidth and congestion are not issues then it is likely that G711 will give the best quality for A VoIP call. Where there are limitations on bandwidth then G729 is probably the best solution.
Experimenting with different codecs will give you the optimum solution based on your network and bandwidth design.