VoIP Bandwidth Calculator
Calculate the real bandwidth a VoIP deployment needs, including the protocol overhead that frequently exceeds the audio payload itself, plus jitter buffer and QoS guidance.
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The overhead is often larger than the audio
G.729 encodes speech at 8 kilobits per second, and a G.729 call over Ethernet consumes about 31 kilobits per second in each direction. The difference is protocol overhead: every packet carries 12 bytes of RTP header, 8 of UDP, 20 of IPv4 and 18 of Ethernet framing, which is 58 bytes wrapped around a 20 byte payload. Sizing a link on the codec rate alone under provisions it by a factor of nearly four, and the resulting call quality problems are usually blamed on the codec.
Packet interval trades bandwidth against latency
Sending a packet every 20 milliseconds is the default because it balances the two. A 30 millisecond interval carries more audio per packet, so the fixed header cost is amortised further and bandwidth drops noticeably, at the cost of 10 milliseconds more latency and a larger loss when a packet goes missing. Below 20 milliseconds the overhead ratio becomes absurd, with more header than payload on every packet.
Latency budget matters more than bandwidth
ITU-T G.114 puts the target for one way mouth to ear delay at under 150 milliseconds, with up to 400 acceptable for some applications. That budget includes codec processing, packetisation, network transit and the jitter buffer at the far end, and the jitter buffer is where most of the controllable delay sits. A link with ample bandwidth and 200 milliseconds of latency produces worse calls than a constrained link with 40, which is why prioritising voice traffic matters more than provisioning for it.
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Frequently Asked Questions
Why does a G.729 call need 31 kbit/s rather than 8?
Because each packet carries 58 bytes of RTP, UDP, IP and Ethernet headers around a 20 byte audio payload. At 50 packets per second that overhead alone is 23 kbit/s. The codec rate describes the audio, not the traffic.
What is RTP header compression?
cRTP compresses the 40 byte RTP, UDP and IP header down to 2 to 4 bytes on a point to point link, which cuts G.729 bandwidth by more than half. It is CPU intensive on the router and only works link by link, so it suits slow WAN circuits and is unnecessary on Ethernet.
How much bandwidth should I reserve for voice?
No more than about a third of the link for the voice class, which is the usual guidance. Beyond that, other traffic is starved during call peaks and the network becomes unusable for everything else while still not guaranteeing voice quality under real congestion.
What latency is acceptable?
ITU-T G.114 targets under 150 ms one way mouth to ear, with up to 400 ms tolerable for some uses. Above about 200 ms conversation becomes stilted as people talk over each other. Jitter should stay under 30 ms and loss under 1 percent for G.711.
Does the jitter buffer add delay?
Yes, directly. It holds packets to smooth out variation in arrival time, so a 60 ms buffer adds 60 ms to the one way delay. An adaptive buffer sizes itself to observed jitter, which is why measuring jitter matters more than provisioning bandwidth.
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How to Use
Choose a codec, packet interval and call count to get the true bandwidth requirement.
Disclaimer: This tool is provided "as is" without warranty of any kind. Results are for educational and utility purposes.
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