Over the past few days, some readers have informed us that they have encountered the Cisco Codec Calculator.
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This document explains the bandwidth of the voice codec and the calculation functions for changing or maintaining bandwidth when using Voice over IP (VoIP). One of the most important factors to consider when building packet voice networks is proper bandwidth planning. As part of the desired bandwidth, bandwidth calculation is an important factor to consider when troubleshooting packet voice networks for very good voice quality.
How is voice bandwidth calculated?
(more precisely) Bandwidth required = Number of simultaneous calls X Size of your voice packets (Kbps, kilobits per second) It’s very simple. If you have any questions, you can download our VoIP Reference Guide or simply call our VoIP experts at 1-800-398-8647.
Note. In addition to this document, you can use any of the TAC Voice Bandwidth Codec Calculator client tools (registered users only). This tool guarantees information on how I can calculate the required bandwidth for packet calls.
VoIP – Bandwidth Per Call
How much bandwidth does a Cisco IP phone use?
On circuit-switched voice mail networks, all voice calls use fixed-bandwidth 64 kbps connections, regardless of how much of the conversation is allocated and how much silence. In VoIP networks, all calls and interrupts are packetized. Silent packets can be removed using voice detection.howling activity (VAD).
40 bytes for IP header (20 bytes) / datagram protocol (UDP) (8 bytes) / transport header preal time (RTP) (12 bytes).
Compressed Real Time Protocol (cRTP), IP/UDP/RTP reduces actual headers to 2 or 4 bytes (cRTP is obviously not available on Ethernet).
6 bytes for Multilink Point-to-Point (MP) and Frame Relay Forum (FRF). Only 12-level heading (L2).
1 byte for specific end-of-frame indicator in MP frames and relay frames.
18 bytes for Ethernet L2 headers containing 4 bytes of frame check sequence (FCS) or cyclic redundancy check (CRC).
Note. This table only contains formulas for standard voice payload sizes in Cisco CallManager gateways or Cisco IOS® H.323 software. For additional calculations related to different Angular payload sizes and other protocols such as Voice over Frame Relay (VoFR) and therefore the use of Voice over ATM (voatm), each TAC Voice Bandwidth Codec Calculator (registered customers only) provides tool.
|Codec information||Bandwidth calculation|
|Codec and bitrate (kbps)||Codec sample size (bytes)||Codec sampling interval (ms)||Mean Opinion Score (MOS)||Voice payload size (in bytes)||Voice payload size (ms)||Packets per second MP (pps)||Bandwidth or FRF.12 (Kbps)||Bandwidth with cRTP or mp FRF.12 (kbps)||Ethernet throughput (kbps)|
|G.711 (64 kbps)||80 bytes||10ms||4.1||160 bytes||20ms||50||82.8 kbps||67.6 kbps||87.2 kbps|
|G.729 (8 kbps)||10 bytes||10ms||3.92||20 bytes||20ms||50||26.8 kbps||11.6 kbps||31.2 kbps|
|G.723.1 (6.3 kbps)||24 bytes||30ms||3.9||24 bytes||30ms||33,3||18.9 kbps||8.8 kbps||21.9 kbps|
|G.723.1 (5.3 kbps)||20 bytes||30ms||3.8||20 bytes||30ms||33,3||17.9 kbps||7.7 kbps||20.8 kbps|
|G.726 (32 kbps)||20 bytes||5 ms||3.85||80 bytes||20ms||50||50.8 kbps||35.6 kbps||55.2 kbps|
|G.726 kbps)||15 (24 bytes||5 ms||20ms||50||42.8kbps||27.6 kbps||47.2 kbps|
|G.728 (16 kbps)||10 bytes||5 ms||3.61||60 bytes||30ms||33,3||28.5 kbps||18.4 kbps||31.5 kbps|
|G722_64k (64 bytes||10 kbps)||80ms||4.13||160 bytes||20ms||50||82.8 kbps||67.6 kbps||87.2 kbps|
|ilbc_mode_20 (15.2 kbps)||38 bytes||20ms||No data||38 bytes||20ms||50||34.0 kbps||18.8 kbps||38.4 kbps|
|ilbc_mode_30 (13.33 kbps)||50 bytes||30ms||No data||50 bytes||30ms||33,3||25.867 kbps||15.73 kbps||28.8 kbps|
|Enabled codec bitrate (kbps)||According to my codec, this is the number of bits per second that needs to be transmitted to transfer a voice call. (codec bitrate codec = sample size and codec sample interval).|
|Codec sample size (bytes)||Depending on the codec, this is the number of bytes captured by the digital signal processor (DSP) in one codec sample interval. For example, the G.729 encoder operates at 10 ms sampling intervals, which corresponds to ten bytes (80 bits) per sample at a bit rate of 8 kbps (codec bit rate codec = chunk size / codec sample interval).< / td >|
|Codec sampling interval (ms)||This is an example iteration where the codec works. For example, the G.729 encoder operates at 10 ms sampling intervals, which is known to be 10 bytes (80 bits) at a sampling rate of 8 kbps. (codec sum of codec bits = sample size per codec sample interval).|
|Mean Opinion Score (MOS)||MOS is a system used for voice quality over telephone connections. On a scale of one (poor) to five (excellent), a variety of MOS listeners rate the quality of a style sample. Medium targets are used to enable MOS for the codec.|
|Voice payload size (in bytes)||The human voice payload size is the number of bytes (or bits) that are typically packed into a packet. The voice payload size must be a multiple of the codec subset size. For example, G.729 packets can certainly use 10, 20, 30, 40, 48, or 60 bytes of payload i.|
|Speech payload size (ms)||Speech analysis payload size can also be represented only as codec samples. For example, a G.729 voice payload power of 20 ms (two examples of 10 Microsoft codecs) represents a human payload of 20 bytes [(20 bytes – 8)/(20 ms) equals 8 kbps]< /td >|
|PPS||PPS is a large number of packets that must be transmitted every second to ensure the codec’s bit rate. For example, a G.729 call with a voice payload size of 20 bytes (160 bits) per packet would transmit 50 packets per second [50 packets per second = (8 kbps) or (160 bits per packet)]|
Total packet size = (L2 headers: MP, FRF.12, or Ethernet) + (IP/UDP/RTP headers) + (voice payload size)
How much bandwidth does a g711 call use?
The most commonly used codec is called G.711 and uses 64 kilobits per second plus additional background data, which can result in 80-90 kbps of IP throughput.
PPS is equal to (codec bit rate) or (voice payload size)
Bandwidth = total packet length and size * PPS
For example, an ordered skipThe th capability for G.729 email (8 kbps codec bit rate) is cRTP, MP, and the default voice payload is twenty:
Which codec is best for VoIP?
You have the ability to use the G.711 codec for all kinds of VoIP applications without paying license fees. There is also no digital compression, so it is probably considered an excellent VoIP codec for connecting to the appropriate public switched telephone network (PSTN).
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