I would like to talk a little bit about the bandwidth requirements for a VoIP network. Bandwidth is a measurement of how much traffic can go through your network.
For lower bandwidth connections (i.e. less than 1Mbps) it is recommended that no more than 50-55% of the total available bandwidth be used for voice traffic. For connections greater than 1 Mbps you can use up to 85% of your available bandwidth for voice traffic. This leaves some headroom on the links to handle bursts of traffic without overloading the link. A bottleneck can occur at any point on the network, including device to switch, switch to switch, switch to gateway, and many more. As such we need to ensure that there is sufficient bandwidth in every pipe on our network to handle the traffic that is being sent.
In addition to the amount of bandwidth required by the data being transmitted, there is additional bandwidth required for every packet sent based on which layer 2 method you are using. For a Frame Relay connection there is a requirement of 6 bytes of overhead. For Point-to-Point Protocol (PPP) there is a requirement for 8 bytes of overhead. Full Duplex Ethernet can be either 38 bytes of 44 bytes – the additional 6 bytes are only used if you have implemented 802.1p, or packet prioritization.
The amount of bandwidth that is required must include the amount of bandwidth required for the CODEC and the amount of bandwidth required for the transport of the packet, including the layer 2 overhead.
Each packet has the following components:
Voice Payload à This is a variable size, and depends on the configuration of your VoIP implementation
RTP Header à 12 bytes, this contains information about the source & destination
UDP Header à 8 bytes, this contains information on the ports used at the source and destination
IPv4 Header à 20 Bytes, included in all IPv4 packets. An IPv6 header is even larger, as the header must include larger addresses and other information
In addition, there are two further components, the Frame Check Sequence of 4 Bytes and the Interframe Gap of 12 Bytes
As we can see, the packet size is 40 bytes plus the size of the payload. As I mentioned earlier, the size of the payload is variable, and depends on the sample length and the CODEC bit rate. The payload size can be calculated using the following formula:
Payload = Bit Rate * Sample Length (s) / 8 bits per byte
This will give us the number of Bytes of payload that needs to be included in the package. We then add to this the amount of overhead from Layer 2 and the amount of IP overhead to give us our total packet size. As an example, let’s take a 20 ms sample size using G.729 (8kbps) over Ethernet using 802.1p:
Payload = 8kbps * 0.02s / 8
Payload = 8000 bps * 0.02s / 8
Payload = 160 bps / 8 = 20 Byte payload size
We then add our overhead to the payload:
Packet size = Payload + Layer 2 overhead + IP Overhead + FCS + IFG
Packet size = 20 Bytes + 44 Bytes + 40 Bytes + 4 Bytes + 12 Bytes
Packet size = 120 Bytes for 20 ms of voice traffic
We can then take this and determine how much bandwidth is required by converting this to bps. We need to determine how many packets per second of voice. As we have seen, 20ms is 0.02 seconds of data, which means that we are using 50 packets per second. So, we multiply our 120 Bytes/packet * 50 packets/second to give us 6000 Bytes per second. We then multiply by 8 to give us bits per second ~ 48000 bps of bandwidth for each direction of a call.
It should also be noted that if we are using the Real-Time Control Protocol we need to add an additional 146 bytes of data to our packet size.
Voice Activity Detection can reduce our bandwidth requirements, as was mentioned before. It enables a data network carrying voice traffic to detect the absence of audio. When audio is absent, it will instead send a signal that indicates that there is no noise, thereby reducing the payload size. A conservative estimate of the bandwidth savings gained using VAD is 30%. VAD is also referred to as Silence Suppression.
EDIT: A Bandwidth calculator is available here.
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