Bandwidth Usage and QoS
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| - | Note that G.722 is a wideband (also known as HD voice) codec and is only fully supported in ipbx 2.1.1 and above. Passthrough between two G.722 capable handsets is still possible though. | + | Note that G.722 is a wideband (also known as HD voice) codec and is only fully supported in ipbx 2.1.1 and above. Passthrough between two G.722 capable handsets is still possible in any ipbx v2 release though. |
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Latest revision as of 20:37, 4 February 2011
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.
Bandwidth Usage
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 is only fully supported in ipbx 2.1.1 and above. Passthrough between two G.722 capable handsets is still possible in any ipbx v2 release 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 |
Softphones
There are two options on how to control QoS and bandwidth usage with softphones. You can either add a secondary NIC (network interface card) to your PC or share the bandwidth with your regular non-VoIP data.
1. If you have a second NIC and appropriately configured, data will travel through the phone network, simply because the softphone will try to register from an IP in the phone network (multi-homed PC). Check your operating system documentation on how to setup multiple network interface cards.
2. Depending on your non-voice network usage, the bandwidth on your network might still be plenty to run softphones. A single call only uses about 0.1% of your 100Mbit LAN. In terms of QoS, softphones such as Counterpath's X-Lite/Eyebeam will do QoS tagging (through the operating system IP stack) and if your Ethernet switch does support QoS tagging (IEEE 802.1p) VoIP data will be prioritized.
