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When calculating bandwidth, one can't assume that every channel is used all the time. Normal conversation includes a lot of silence, which often means no packets are sent at all. So even if one voice call sets up two 64 Kbit RTP streams over UDP over IP over Ethernet (which adds overhead), the full bandwidth is not used at all times.
A codec that sends a 64kb stream results in a much larger IP network stream. The main cause of the extra bandwidth usage is IP and UDP headers. VoIP sends small packets and so, many times, the headers are actually much larger than the data part of the packet.
IAX2 trunking helps with the IP overhead, but only when you are sending more than 2 or so calls between the same Asterisk servers. John Todd has done some useful practical testing, named IAX2 trunking: codec bandwidth comparison notes and results.
The bandwidth used depends also on the datalink (layer2) protocols. Several things influence the bandwidth used, payload size, ATM cell headers, VPN headers, use of header compression and IAX2 Trunked. You can see the influence of some of this factors using the Asteriskguide bandwidth calculator.
Teracall has the table which shows how the codec's theoretical bandwidth usage expands with UDP/IP headers:
Codec BR 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
BR = Bit rate
NEB = Nominal Ethernet Bandwidth (one direction)
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- Asterisk guide bandwidth calculator, it calculates 32xCodecs, 13xL2-Protocols, IAX Trunked, ATM, cRTP, VPN
- Asteriskguru's bandwidth calculator
- Bidirectional Throughput VoIP Test - Utility to test if your network is ready for VoIP Service - Throughput, Capacity, Latency, and SIP & MGCP Port Tests
- Cisco.com whitepaper: Per call bandwidth consumption This Cisco whitepaper explains how you can calculate bandwidth with VOIP.
- Erlang.com online bandwidth calculators: http://www.erlang.com/calculator/
- IAX2 trunking: codec bandwidth comparison notes and results
- IPhone App: VoIP Bandwidth Calc and app store link
- Android App: VoIP Bandwidth Calc
- Newport Networks Online VoIP Bandwidth Calculator and White Paper
- Packetizer.com: Bandwidth calculators
- Speex.org: Codec feature comparisons
- VoIP traffic bandwidth calculator This bandwidth tool support many codec (g711, g729, g723_6.3Kbps, g723_5.3Kbps, g726_32Kbps, g726_24Kbps, g728_16Kbps, g722, ilbc 15.2Kbps, ilbc 13.33Kbps), many signaling protocol (sip, mgcp, h323, iax), transport protocol (RTP, cRTP), layer 2 protocol (Ethernet or non ethernet)
- VoIP bandwidth calculator designed by network engineer Supports many signaling protocol (sip, mgcp, h323, iax), L2, L3, tunnelings, codecs. Universal app so works on iphone/ipod/ipad.
'Optima Saver' is a popular bandwidth optimization service for VOIP traders. This service is introduced by ImproLabs Pte Ltd, a Singapore based IT & communications solutions Company. Optima Saver can run upto 5 times more calls.
- Reduces bandwidth consumption for VOIP call termination upto 80%.
- Supports major audio codecs.
- Works with all major SIP termination gateways.
- Full featured web based platform is integrated.
- Compatible with different kinds of internet networks.
- Compatible with public, private, static, DHCP networks.
- It helps to by-pass voice port blocking problem.
- Works from behind the NAT and firewall.
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