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Recently, some of our users have encountered a known bug with the Cisco Voice Bandwidth Codec Calculator. Several factors can cause this problem. Let’s take a look below.
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This document explains the bandwidth calculations of the voice codec and features to change bandwidth or simply save money when using Voice over IP (VoIP). One of the most important factorsThe things to consider when building voice networks is proper bandwidth planning. When planning bandwidth, calculating bandwidth is an important factor to consider when designing packet voice networks and eliminating the resulting errors in order to achieve good voice quality.
Note. In addition to this document, you can use the TAC Voice Bandwidth Codec Calculator (tool for registered customers only). This tool provides information on quantifying the bandwidth required to make batch calls.- Pro
VoIP Call Bandwidth
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40 bytes for IP (20 bytes) / User Datagram Protocol (UDP) (8 bytes) / Real Time Transport Protocol (RTP) header (12 bytes).
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Compressed Real Time Protocol (cRTP) reduces the IP / UDP / RTP headers to 2 or bytes (cRTP is not available on Ethernet).
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6 bytes for multipoint protocolzi (MP) or Frame Relay Forum (FRF). 12 Layer 2 (L2) header. byte
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1 for end of frame flag for MP and Frame Relay frames.
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18 bytes for Ethernet L2 header, including 4 bytes related to frame sequence check (FCS) or cyclic redundancy check (CRC). This
Note. The table contains only rule sets for standard voice load sizes that can be found in Cisco CallManager or Cisco IOS ® H.323 gateways. For additional calculations involving many types of voice payload sizes and other protocols used, such as Voice over Frame Relay (VoFR) rather than Voice over ATM (VoATM), use the Voice tac Bandwidth Codec Calculator (for registered subscribers) unambiguously).
Codec info colspan = “4” | bandwidth calculation | ||||||||
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Codec and bit rate (Kbps) | Approximate codec size (bytes) | Codec sampling interval (ms) | Average Opinion Score (MOS) | Voice load size (bytes) | Voice load size ki (ms) | Packets Per Second (PPS) | Bandwidth MP or FRF.12 (Kbps) | Bandwidth with cRTP MP FRF ou.12 (Kbps) | Ethernet Bandwidth (Kbps) |
G.711 (64 kbps) | 80 bytes | 10 ms | 4.1 | 160 bytes | 20 ms | 50 | 82.8 kbps | 67.6 kbps | 87.2 kbps |
G.729 (8Kbps) | 10 bytes | 10 ms | 3.92 | 20 bytes | 20 ms | 50 | 26.8 kbps | 11.6 kbps | 31.2 kbps |
G.723.1 (6.3 Kbps) | 24 bytes | 30 ms | 3.9 | 24 bytes | 30 ms | 33.3 | 18.9 Kbps | 8.8 kbps | 21.9 kbps |
G.723.1 (5.3 Kbps) | 20 bytes | 30 ms | 3.8 | 20 bytes | 30 ms | 33.3 | 17.9 kbps | 7.7 kbps | 20.8 kbps |
G.726 (32 Kbps) | 20 bytes | 5 ms | 3.85 | 80 bytes | 20 ms | 50 | 50.8 kbps | 35.6 kbps | 55.2 kbps |
G.726 (24 kbps) | 15 women | 20 bytes | 5 ms | 50 | 42.8 kbps | 27.6 kbps | 47.2 Kbps | ||
G.728 (16Kbps) | 10 bytes | 5 ms | 3.61 | 60 bytes | 30 ms | 33.3 | 28.5 Kbps | 18.4 kbps | 31.5 kbps |
G722_64k (64 kbps) | 80 women | 4 bytes | 10.13 | 160 bytes | 20 ms | 50 | 82.8 kbps | 67.6 kbps | 87.2 kbps |
ilbc_mode_20 (15.2 Kbps) | 38 bytes | 20 ms | no data | 38 bytes | 20 ms | 50 | 34.0 Kbps | 18.8 Kbps | 38.4 kbps |
ilbc_mode_30 (13.33 Kbps) | 50 bytes | 30 ms | no data | 50 bytes | 30 ms | 33.3 | 25 867 Kbps | 15.73 kbps | 28.8 kbps |
Explanation Of Terms
Codec Bit (Kbps) | Depending on the codec, this is the number of bits per second that must be transmitted to advance a voice call. Bit (codec rate means codec sample size / codec structure interval). |
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Approximate codec size (bytes) | Based on the codec until the number of bytes captured by the digital signal processor (DSP) at each individual codec sampling interval is reached. For example, a G.729 encoder operates at sampling intervals of 10 ms, which corresponds to 10 bytes (80 dfor each individual bit) with a bit cycle of 8 kbps. Bit (codec rate is the sample size of the codec / codec bit of the music interval). Sample |
Codec Interval (ms) | This is the sampling period in which the codec runs. As a demonstration, the G.729 encoder operates with 10 ms samples, which corresponds to ten bytes (80 per bit) of sample at each 8 kbps rate. Bit (codec rate = codec sample size or codec sample interval). |
Average Opinion Score (MOS) | MOS is a complete used system for exceptional voice assessment of telephone connections. With MOS, a wide range of listeners rate voice quality on a scale of one (poor) to five (excellent). The scores are averaged to provide our own MOS for the codec. |
Voice load size (bytes) | The size of the voice payload, which is the number of bytes (or bits) filled in a packet. The size of the voice payload must be a multiple of the selected codec size. For example, G.729 packages can make full use ofUse 10, 20, 30, 40, 50, 60, or only bytes of the voice payload. |
Voice load size (ms) | The size of the voice payload can also be better represented than the sample codecs. For example, a G.729 voice payload metric of 20 ms (two Microsoft codec samples of 10 each) represents a 60 byte voice payload [(20 bytes * 8) – (20 ms) = 8 Kbps] |
PPS | PPS is the number of packets that must be transmitted every second to ensure the bit rate of the codec. For example, when calling G.729 with a voice payload size of 20 (160 bits per packet) per packet, 50 packets must be sent every second [50 packets per second = (8 Kbps) / (160 bits per packet)] |
Bandwidth Formulas
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Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP / UDP / RTP header) + (voice load size)
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PPS = (Minor Codec Rate) / (Voice Load Size)
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Total bandwidth assumes * PPS packet size
Calculation Example
NapFor example, the total bandwidth required for G.729 is accounted for (8Kbps codec) with cRTP, MP and a standard 20-byte stride payload:
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Total packet size = (bytes) (6 byte MP h2 tags) + (2 bytes collapsed IP / UDP / RTP header) + (20 bytes voice payload) = twenty-eight bytes
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Total packet size (bits) = (28 bytes) 3.8 bits per byte = 224 bits
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PPS matches (codec bit rate 8 kbps) versus (160 bits) = 50 pps
Note: 160 elements = 20 bytes (standard voice data) 8 bits per byte
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Bandwidth per call corresponds to the size of a voice packet (224 bits) * 60 pps = 11.2 Kbps
Configuring Voice Payload Sizes In Cisco CallManager And Cisco IOS Voice Gateways
The payload per packet size can be specifically configured in Cisco CallManager and Cisco IOS gateways.
Note. If a Cisco IOS gateway is frequently configured in Cisco CallManager as bearer, your own gateway control protocol (MGCP), all codec information (codec type, payload size, detection signal andactivity, etc.) will be tracked by Ci.
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Total Block Size = (L2 Header: MP or FRF.12 or Ethernet) + (IP / UDP / RTP Header) + (Voice Load Size)PPS = (codec bitrate) / (voice load size)Throughput = total container size * PPS.
VoIP document consumption ranges from 0.5 megabytes (MB) per minute of call (with G.729 codec) to 1.3 MB / minute with G.711. This number fluctuates depending on the further development of the VoIP operator and the use of the caller. habit.
711 is a narrowband stereo codec originally developed for use in telephone systems, providing high quality audio at 64 kbps. The G. 711 transmits audio signals in most of the 300-3400 Hz bands and samples today’s people at 8000 samples per second with a tolerance of 50 parts per million (ppm) in price.