Difference between revisions of "Bandwidth estimates"
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* [http://www.bandcalc.com/ Packetize VoIP Bandwidth Calculator] - Audio codecs only, support for multiple channels. | * [http://www.bandcalc.com/ Packetize VoIP Bandwidth Calculator] - Audio codecs only, support for multiple channels. | ||
* [http://www.erlang.com/calculator/eipb/ Erlangs and VoIP Bandwidth Calculator] - Audio codecs only, uses "Erlangs" as parameter. (''Erlangs'' is the number of hours of call traffic there are during the busiest hour of operation of a telephone system.) | * [http://www.erlang.com/calculator/eipb/ Erlangs and VoIP Bandwidth Calculator] - Audio codecs only, uses "Erlangs" as parameter. (''Erlangs'' is the number of hours of call traffic there are during the busiest hour of operation of a telephone system.) | ||
− | + | * [http://www.erlang.com/calculator/lipb/ Lines to VoIP Bandwidth calculator] - estimate the bandwidth required to transport a given number of voice paths through an IP based network. Reverse calculations are also possible. | |
== Audio codecs == | == Audio codecs == |
Revision as of 23:07, 30 January 2007
Packet overhead
Communication over an IP-network has some overhead for each data-packet. This overhead can be seen as a constant value in calculations, although it is dependant on a few parameters like the networkstructure and wether it is an TCP or UDP packet.
For TCP, an overhead of 52 Bytes is average and for UDP it is 28 Bytes.
See for more detailed information ethernet payload datarates.
Along with the data used by the codec(s), some control data also has to be transmitted.
This means the overhead per packet is even greater than only the IP-headers.
This overhead is different for several protocols (e.g. SIP and IAX) and may also be dependant on the number of streams combined in the packet. (e.g. audio and video or even multiple conversations)
Trunking
Because of this packet-overhead, several protocols (like IAX2) support some kind of multiplexing. This means in each datapacket which is sent over the network, several packets of multiple conversations are put together in order to minimize the amount of packets on the network, thus maximizing the payload and minimizing the overhead.
This means the bandwith used between 2 PBX stations can be seen as:
- BRcodec * #channels + IP-overhead + PBX-protocol-overhead.
This saves (n-1) * (IP-overhead + PBX-overhead) in traffic and even more important, all traffic can be sent over one connection.
Bandwidth calculators
- Packetize VoIP Bandwidth Calculator - Audio codecs only, support for multiple channels.
- Erlangs and VoIP Bandwidth Calculator - Audio codecs only, uses "Erlangs" as parameter. (Erlangs is the number of hours of call traffic there are during the busiest hour of operation of a telephone system.)
- Lines to VoIP Bandwidth calculator - estimate the bandwidth required to transport a given number of voice paths through an IP based network. Reverse calculations are also possible.
Audio codecs
Bandwidth Requirements for Several Common VoIP Compression Algorithms
Codec | Bit rate | NEB |
---|---|---|
G.711 | 64 Kbps | 87.2 Kbps |
G.729 | 8 Kbps | 31.2 Kbps |
G.723.1 | 6.4 Kbps | 21.9 Kbps |
G.723.1 | 5.3 Kbps | 20.8 Kbps |
G.726 | 32 Kbps | 55.2 Kbps |
G.726 | 24 Kbps | 47.2 Kbps |
G.728 | 16 Kbps | 31.5 Kbps |
iLBC | 15 Kbps | 27.7 Kbps |
NEB = Nominal Ethernet Bandwidth (one direction)
- G711u/a - Requires 64+Kbps (inbound and outbound) continuous data stream for a quality call, plus the SIP signal. A 128Kbps connection has shown to be the minimal connection needed in our testing.
- G723 - Requires 6.4+Kbps (I and O) continuous data stream and also needs room for the SIP signal, 28.8Kbps connection has shown to be the minimal connection.
(Source)