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RTP Packet Rate Calculator

Networking
Default: 2000 kbps
Default: 33 ms
IPv6 adds 20 bytes vs IPv4
0 = no MTU limit

Include Ethernet Framing (LAN / 46 B extra)

Total Bandwidth

2.021 Mbps

1 stream

PPS per Stream

30.3 pps

33 ms frame interval

Total PPS

30.3 pps

across 1 stream

Header Overhead

1.0%

86 B per packet

Payload vs. Protocol Overhead (per packet)

Overhead (1.0%)

Payload (99.0%)

Overhead (86 B)

Payload (8250 B)

Protocol Header Breakdown

LayerBytes% of Wire
RTP Header12 B

0.1%

UDP Header8 B

0.1%

IPv4 Header20 B

0.2%

Ethernet Frame + Wire46 B

0.6%

RTP Payload8250 B

99.0%

Total Wire8336 B100%

Full Results Summary

MetricValue
Codec bitrate2.000 Mbps
Bandwidth per stream (with headers)2.021 Mbps
Total bandwidth (1 stream)2.021 Mbps
Total bandwidth (kbps)2020.85 kbps
Total bandwidth (Mbps)2.0208 Mbps
PPS per stream30.3 pps
Total PPS (1 stream)30.3 pps
RTP timestamp increment2,970

About This Tool

RTP Packet Rate Calculator – Bandwidth and Overhead for Streaming

The RTP Packet Rate Calculator helps engineers, developers, and network architects compute the exact number of packets per second (PPS) and total bandwidth consumed by real-time audio and video streams over IP networks. It accounts for the full protocol stack — RTP, UDP, IP, and optional Ethernet framing — so you can size links, plan capacity, and diagnose overhead issues with precision.

What Is RTP and Why Does Packet Rate Matter?

RTP (Real-time Transport Protocol, RFC 3550) is the standard transport for audio and video over IP. It runs over UDP and adds a 12-byte header carrying sequence numbers, timestamps, and synchronisation source (SSRC) identifiers. RTP is used in WebRTC, SIP, RTSP, live-streaming pipelines, and conferencing systems.

Packet rate matters for two reasons. First, total PPS determines the forwarding load on routers and switches — a device can run out of packets-per-second capacity long before saturating raw bandwidth. Second, protocol headers add overhead to every packet; for low-bitrate audio codecs this overhead can exceed the payload, dramatically increasing the real bandwidth requirement.

How Packet Rate Is Calculated

The tool derives PPS from two inputs: the codec bitrate and the frame (packetization) interval.

  • Payload bytes per packet = (bitrate_bps ÷ 8) × (frame_ms ÷ 1000)
  • PPS per stream = 1000 ÷ frame_ms (or multiplied by fragment count when MTU fragmentation is enabled)
  • Wire bytes per packet = payload + RTP (12 B) + UDP (8 B) + IP (20 B IPv4 / 40 B IPv6) + optional Ethernet (46 B)
  • Total bandwidth = wire bytes × 8 × PPS × number of streams

The overhead percentage shows how much of the total wire bandwidth is consumed by protocol headers rather than useful media content.

Supported Codecs

The calculator includes common audio and video codecs with their typical bitrates and default frame intervals:

  • Audio: Opus (48 kbps and 128 kbps), AAC (128 / 256 kbps), G.711u (64 kbps), G.729 (8 kbps).
  • Video: H.264 (500 kbps SD, 2 Mbps HD 720p, 4 Mbps HD 1080p), H.265/HEVC (1.5 Mbps), VP8 (1 Mbps), VP9 (2 Mbps), AV1 (1.5 Mbps).
  • Custom: Enter any bitrate and frame interval for non-listed codecs or proprietary streams.

Selecting a preset codec automatically populates the bitrate and frame interval fields with typical values; you can override either field.

MTU Fragmentation

On standard Ethernet paths with a 1500-byte MTU, a large video frame may exceed the maximum allowed RTP payload size. Enter a Max RTP Payload value (for example 1452 bytes for IPv4 over Ethernet) to simulate fragmentation. The calculator splits the frame into multiple RTP packets and recalculates PPS and bandwidth to reflect the true forwarding load — which can be significantly higher than the unfragmented estimate.

IPv4 vs IPv6 Impact

IPv6 headers are 40 bytes versus 20 bytes for IPv4, adding 20 extra bytes per packet. For high-PPS audio streams (e.g. 50 pps per stream), this adds 20 × 50 × 8 = 8 kbps per stream in additional overhead — a negligible amount for high-bitrate video but meaningful when sizing a large number of low-bitrate audio channels.

RTP Timestamp Increment

The results include the RTP timestamp increment per packet. This is the number of clock ticks the RTP timestamp advances between consecutive packets, computed as clock_rate × (frame_ms ÷ 1000). For video codecs using a 90 kHz clock and 33 ms frames (30 fps), the increment is 2970. This value is required when manually crafting RTP streams or debugging timestamp drift in streaming pipelines.

Practical Uses

  • Link sizing: calculate the bandwidth budget for a given number of simultaneous video conferencing sessions.
  • Router capacity planning: confirm that PPS requirements are within the forwarding engine limits of edge or core routers.
  • Codec selection: use the Codec Comparison tab to compare overhead percentage and total bandwidth across all supported codecs at once.
  • MTU optimisation: identify whether fragmentation is causing a PPS spike on constrained paths and determine the optimal maximum payload size.

Frequently Asked Questions

Is the RTP Packet Rate Calculator free?

Yes, RTP Packet Rate Calculator is totally free :)

Can I use the RTP Packet Rate Calculator offline?

Yes, you can install the webapp as PWA.

Is it safe to use RTP Packet Rate Calculator?

Yes, any data related to RTP Packet Rate Calculator only stored in your browser (if storage required). You can simply clear browser cache to clear all the stored data. We do not store any data on server.

How does the RTP Packet Rate Calculator work?

Select a codec (or enter a custom bitrate), set the frame/packetization interval, number of simultaneous streams, and transport options. The tool computes payload bytes per RTP packet, packets per second (PPS), total bandwidth including all protocol headers (RTP, UDP, IP, optional Ethernet), and the overhead percentage.

What is an RTP packet and why is the packet rate important?

RTP (Real-time Transport Protocol, RFC 3550) wraps audio or video payloads for real-time delivery over IP networks. Packet rate (PPS) determines CPU and forwarding load on network devices — high PPS from many short-interval streams can saturate a router's packet processing engine even when total bandwidth is low.

How does the frame interval (ptime) affect bandwidth and packet rate?

A shorter frame interval produces more packets per second, each carrying a smaller payload, which increases protocol overhead as a percentage of total bandwidth. A longer interval reduces PPS and overhead but increases latency and jitter sensitivity. For audio, 20 ms is the common default; for video, 33 ms corresponds to 30 fps.

What happens when I set a maximum RTP payload size (MTU constraint)?

If the payload computed from bitrate and frame interval exceeds the specified maximum, the tool fragments the frame into multiple RTP packets. This simulates MTU-limited paths (e.g. 1500-byte Ethernet MTU minus IP/UDP/RTP headers leaves roughly 1452 bytes for payload in IPv4). Fragmentation multiplies PPS and total bandwidth accordingly.

Why is the actual bandwidth higher than the codec bitrate?

Every RTP packet carries at minimum 40 bytes of protocol overhead: 12 bytes RTP + 8 bytes UDP + 20 bytes IPv4. For low-bitrate audio codecs (e.g. G.729 at 8 kbps), headers can represent more than 50% of the total wire bandwidth. Adding Ethernet framing adds another 46 bytes per packet.

What is the RTP timestamp increment shown in the results?

RTP timestamps advance by a fixed number of clock ticks per packet, derived from the codec's clock rate and the frame interval. For example, with a 90 kHz video clock and 33 ms frames, the timestamp increments by 2970 ticks per packet. This value is used when manually constructing or debugging RTP streams.