Bandwidth Usage and QoS
From Taridium
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| - | How much Internet bandwidth do I need for my ipbx calls? This will depend on the codec you are using. The standard codec for ipbx is G.711 ( | + | How much Internet bandwidth do I need for my ipbx calls? This will depend on the codec you are using. The standard codec for ipbx is G.711 (µLaw/aLaw) and this is the recommended codec for use in your local area network. |
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| + | µLaw (pronounce mu-law, the µ is a Greek mu) and A-Law operate in the same way; that is they both sample audio at 8KHz and taking into account low frequency and anti-alias filters give the standard telephony frequency response of 300Hz to 3400Hz. | ||
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| + | In aLaw the audio is sampled with a 13 bit resolution and µLaw with 14 bit resolution. These samples are then logarithmically compressed so that in both cases the amplitude is reduced to 8 bits. 8 bits at 8KHz results in 64kbit bandwidth. | ||
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| + | If you are using call termination to the PSTN in Europe, Asia and Australia it makes often sense to change your defaults to aLaw within your VoIP network since this is the codec that is used on ISDN lines and with other termination providers. | ||
When calculating the actual amount of bandwidth, you must take into consideration that every RTP packet has an IP header in addition to the actual data (payload); this header must be transferred along with the data, and therefore requires a certain amount of bandwidth. Since RTP (voice stream) makes use of many small packets, the amount of bandwidth used by IP headers becomes unexpectedly significant. See the table below for estimated | When calculating the actual amount of bandwidth, you must take into consideration that every RTP packet has an IP header in addition to the actual data (payload); this header must be transferred along with the data, and therefore requires a certain amount of bandwidth. Since RTP (voice stream) makes use of many small packets, the amount of bandwidth used by IP headers becomes unexpectedly significant. See the table below for estimated | ||
Revision as of 14:12, 29 September 2009
How much Internet bandwidth do I need for my ipbx calls? This will depend on the codec you are using. The standard codec for ipbx is G.711 (µLaw/aLaw) and this is the recommended codec for use in your local area network.
µLaw (pronounce mu-law, the µ is a Greek mu) and A-Law operate in the same way; that is they both sample audio at 8KHz and taking into account low frequency and anti-alias filters give the standard telephony frequency response of 300Hz to 3400Hz.
In aLaw the audio is sampled with a 13 bit resolution and µLaw with 14 bit resolution. These samples are then logarithmically compressed so that in both cases the amplitude is reduced to 8 bits. 8 bits at 8KHz results in 64kbit bandwidth.
If you are using call termination to the PSTN in Europe, Asia and Australia it makes often sense to change your defaults to aLaw within your VoIP network since this is the codec that is used on ISDN lines and with other termination providers.
When calculating the actual amount of bandwidth, you must take into consideration that every RTP packet has an IP header in addition to the actual data (payload); this header must be transferred along with the data, and therefore requires a certain amount of bandwidth. Since RTP (voice stream) makes use of many small packets, the amount of bandwidth used by IP headers becomes unexpectedly significant. See the table below for estimated bandwidth usage.
Note that G.722 is a wideband (also known as HD voice) codec and will only be fully supported in ipbx 2.1. Passthrough between two G.722 capable handsets is still possible though.
| Codec | Bit Rate | Nominal Bandwidth |
|---|---|---|
| G.711 | 64 kbit/s | 87.2 kbit/s |
| G.729 | 8 kbit/s | 31.2 kbit/s |
| G.723.1 | 6.4 kbit/s | 21.9 kbit/s |
| G.723.1 | 5.3 kbit/s | 20.8 kbit/s |
| G.722 | 48 kbit/s | 71.2 kbit/s |
| G.722 | 56 kbit/s | 79.2 kbit/s |
| G.722 | 64 kbit/s | 87.2 kbit/s |
