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Contents
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Introduction
document.
Prerequisites
      Requirements
      Components Used Excellent

      Conventions Good

VoIP – Per Call Bandwidth Average
      Explanation of Terms Fair
      Bandwidth Calculation Formulas
Poor
      Sample Calculation
Configuring Voice Payload Sizes in Cisco CallManager and IOS Gateways
Impact of Changing Voice Payload Sizes
Voice Activity Detection This document
RTP Header-Compression or Compressed RTP (cRTP) solved my
      Heuristics for Compression problem.
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Introduction
One of the most important factors to consider when you build packet voice networks is proper capacity Suggestions for
planning. Within capacity planning, bandwidth calculation is an important factor to consider when you improvement:
design and troubleshoot packet voice networks for good voice quality.

This document explains voice codec bandwidth calculations and features to modify or conserve
bandwidth when Voice over IP (VoIP) is used.

Note: As a complement to this document, you can use the TAC Voice Bandwidth Codec Calculator (  (256 character limit)
registered customers only ) tool. This tool provides information on how to calculate the bandwidth required
for packet voice calls. Send

Prerequisites
Requirements
There are no specific requirements for this document.

Components Used
This document is not restricted to specific software and hardware versions.
Conventions

Refer to Cisco Technical Tips Conventions for more information on document conventions.

VoIP – Per Call Bandwidth

These protocol header assumptions are used for the calculations:

40 bytes for IP (20 bytes) / User Datagram Protocol (UDP) (8 bytes) / Real-Time Transport Protocol (RTP) (12 bytes)
headers.

Compressed Real-Time Protocol (cRTP) reduces the IP/UDP/RTP headers to 2or 4bytes (cRTP is not available over
Ethernet).

6 bytes for Multilink Point-to-Point Protocol (MP) or Frame Relay Forum (FRF).12 Layer 2 (L2) header.

1 byte for the end-of-frame flag on MP and Frame Relay frames.

18 bytes for Ethernet L2 headers, including 4 bytes of Frame Check Sequence (FCS) or Cyclic Redundancy Check
(CRC).

Note: This table only contains calculations for the default voice payload sizes in Cisco CallManager or Cisco IOS® Software
H.323 gateways. For additional calculations, including different voice payload sizes and other protocols, such as Voice over
Frame Relay (VoFR) and Voice over ATM (VoATM), use the TAC Voice Bandwidth Codec Calculator ( registered customers
only ) tool.

Note: 

Codec Information Bandwidth Calculations

Bandwidth
Codec Codec Mean Voice Voice Packets Bandwidth
w/cRTP Bandwidth
Codec & Bit Rate Sample Sample Opinion Payload Payload Per MP or
MP or Ethernet
(Kbps) Size Interval Score Size Size Second FRF.12
FRF.12 (Kbps)
(Bytes) (ms) (MOS) (Bytes) (ms) (PPS) (Kbps)
(Kbps)

80 160
G.711 (64 Kbps) 10 ms 4.1 20 ms 50 82.8 Kbps 67.6 Kbps 87.2 Kbps
Bytes Bytes

10 20
G.729 (8 Kbps) 10 ms 3.92 20 ms 50 26.8 Kbps 11.6 Kbps 31.2 Kbps
Bytes Bytes

24 24
G.723.1 (6.3 Kbps) 30 ms 3.9 30 ms 33.3 18.9 Kbps 8.8 Kbps 21.9 Kbps
Bytes Bytes

20 20
G.723.1 (5.3 Kbps) 30 ms 3.8 30 ms 33.3 17.9 Kbps 7.7 Kbps 20.8 Kbps
Bytes Bytes
 20 80
G.726 (32 Kbps) 5 ms 3.85 20 ms 50 50.8 Kbps 35.6 Kbps 55.2 Kbps
Bytes Bytes

15 60
G.726 (24 Kbps) 5 ms   20 ms 50 42.8 Kbps 27.6 Kbps 47.2 Kbps
Bytes Bytes

10 60
G.728 (16 Kbps) 5 ms 3.61 30 ms 33.3 28.5 Kbps 18.4 Kbps 31.5 Kbps
Bytes Bytes

80 160
G722_64k(64 Kbps) 10 ms 4.13 20 ms 50 82.8 Kbps 67.6Kbps 87.2 Kbps
Bytes Bytes

38 38
ilbc_mode_20(15.2Kbps) 20 ms NA 20 ms 50 34.0Kbps 18.8 Kbps 38.4Kbps
Bytes Bytes

50 50 25.867
ilbc_mode_30(13.33Kbps) 30 ms NA 30 ms 33.3 15.73Kbps 28.8 Kbps
Bytes Bytes Kbps

Explanation of Terms

Codec
Based on the codec, this is the number of bits per second that
Bit
need to be transmitted to deliver a voice call. (codec bit rate =
Rate
codec sample size / codec sample interval).
(Kbps)

Based on the codec, this is the number of bytes captured by the

Codec Digital Signal Processor (DSP) at each codec sample interval.
Sample For example, the G.729 coder operates on sample intervals of
Size 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit
(Bytes) rate of 8 Kbps. (codec bit rate = codec sample size / codec
sample interval).

This is the sample interval at which the codec operates. For

Codec
example, the G.729 coder operates on sample intervals of 10
Sample
ms, corresponding to 10 bytes (80 bits) per sample at a bit rate
Interval
of 8 Kbps. (codec bit rate = codec sample size / codec sample
(ms)
interval).

MOS is a system of grading the voice quality of telephone

connections. With MOS, a wide range of listeners judge the
MOS quality of a voice sample on a scale of one (bad) to five
(excellent). The scores are averaged to provide the MOS for
the codec.
 Voice The voice payload size represents the number of bytes (or bits)
Payload that are filled into a packet. The voice payload size must be a
Size multiple of the codec sample size. For example, G.729 packets
(Bytes) can use 10, 20, 30, 40, 50, or 60 bytes of voice payload size.

Voice The voice payload size can also be represented in terms of the
Payload codec samples. For example, a G.729 voice payload size of 20
Size ms (two 10 ms codec samples) represents a voice payload of
(ms) 20 bytes [ (20 bytes * 8) / (20 ms) = 8 Kbps ]

PPS represents the number of packets that need to be

transmitted every second in order to deliver the codec bit rate.
PPS For example, for a G.729 call with voice payload size per
packet of 20 bytes (160 bits), 50 packets need to be transmitted
every second [50 pps = (8 Kbps) / (160 bits per packet) ]

Bandwidth Calculation Formulas

These calculations are used:

Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size)

PPS = (codec bit rate) / (voice payload size)

Bandwidth = total packet size * PPS

Sample Calculation

For example, the required bandwidth for a G.729 call (8 Kbps codec bit rate) with cRTP, MP and the default 20 bytes of
voice payload is:

Total packet size (bytes) = (MP header of 6 bytes) + ( compressed IP/UDP/RTP header of 2 bytes) + (voice payload of
20 bytes) = 28 bytes

Total packet size (bits) = (28 bytes) * 8 bits per byte = 224 bits

PPS = (8 Kbps codec bit rate) / (160 bits) = 50 pps

Note: 160 bits = 20 bytes (default voice payload) * 8 bits per byte

Bandwidth per call = voice packet size (224 bits) * 50 pps = 11.2 Kbps

Configuring Voice Payload Sizes in Cisco CallManager and IOS Gateways

The voice payload size per packet can be configured in Cisco CallManager and Cisco IOS gateways.

Note: If the Cisco IOS gateway is configured in Cisco CallManager as a Media Gateway Control Protocol (MGCP) gateway,
all the codec information (codec type, payload size, voice activity detection, and so on) is controlled by Cisco CallManager.

In Cisco CallManager, the voice payload size per packet is configurable on a systemwide basis. This attribute is set in Cisco
CallManager Administration (Service > Service Parameters > select_server > Cisco CallManager) with these three service
parameters:

PreferredG711MillisecondPacketSize—(Default setting: 20 ms. Available settings: 10, 20, and 30 ms.)

PreferredG729MillisecondPacketSize—(Default setting: 20 ms. Available settings: 10, 20, 30, 40, 50, and 60 ms.)

PreferredG723MillisecondPacketSize—(Default setting: 30 ms. Available settings: 30 and 60 ms.)

In Cisco CallManager, the voice payload size is configured in terms of milliseconds (ms) samples. Based on the codec, this
table maps some ms samples to the actual payload size in bytes.

Voice Voice
Payload Payload
Codec Comments
Size Size
(ms) (Bytes)

20 ms 160
(default) Bytes
G.711
240
30 ms
Bytes
Notice that the codec bit rate is always
maintained. For example: G.711 codec =
20 ms
20 Bytes [240 bytes * 8(bits/bytes)] / 30 ms = 64
(default)
Kbps
G.729

30 ms 30 Bytes

30 ms
G.723
(default)

In Cisco IOS gateways, a feature is added in Cisco IOS Software Release 12.0(5)T that allows the voice payload size (in
bytes) for VoIP packets to be changed through the Command-Line Interface (CLI). The new command syntax follows:
Cisco-Router(config-dial-peer)#codec g729r8 bytes ?
Each codec sample produces 10 bytes of voice payload.
Valid sizes are:
10, 20, 30, 40, 50, 60, 70, 80, 90, 100, 110, 120,
130, 140, 150, 160, 170, 180, 190, 200, 210, 220, 230
Any other value within the range will be rounded down to nearest valid size.
<10-230> Choose a voice payload size from the list above

Impact of Changing Voice Payload Sizes

The number of codec samples per packet is another factor that determines the bandwidth and delay of a VoIP call. The codec
defines the size of the sample, but the total number of samples placed in a packet affects how many packets are sent per
second.
When you increase the voice payload size the VoIP bandwidth reduces and the overall delay increases. The following
example illustrates this:

G.729 call with voice payload size of 20 bytes (20 ms): (40 bytes of IP/UDP/RTP headers + 20 bytes voice payload)* 8
bits per byte * 50 pps = 24 Kbps

G.729 call with voice payload size of 40 bytes (40 ms): (40 bytes of IP/UDP/RTP headers + 40 bytes voice payload) *
8 bits per byte * 25 pps = 16 Kbps

Note: L2 headers are not considered in this calculation.

Note: The calculations show that while the payload size is doubled, the number of packets per second required is
subsequently cut in half.

Note: As defined in the International Telecommunication Union Telecommunication Standardization Sector (ITU-T) G.114
specifications, the recommended one-way overall delay for voice is 150 ms. For a private network, 200 ms is a reasonable
goal, and 250 ms must be the maximum.

Voice Activity Detection

With circuit-switched voice networks, all voice calls use 64 Kbps fixed-bandwidth links regardless of how much of the
conversation is speech and how much is silence. With VoIP networks, all conversation and silence is packetized. With Voice
Activity Detection (VAD), packets of silence can be suppressed.

Over time and as an average on a volume of more than 24 calls, VAD can provide up to a 35 percent bandwidth savings. The
savings are not realized on every individual voice call, or on any specific point measurement. For the purposes of network
design and bandwidth engineering, VAD must not be taken into account, especially on links that carry fewer than 24 voice
calls simultaneously. Various features such as music on hold and fax render VAD ineffective. When the network is
engineered for the full voice call bandwidth, all savings provided by VAD are available to data applications.

VAD also provides Comfort Noise Generation (CNG). Because you can mistake silence for a disconnected call, CNG
provides locally generated white noise so the call appears normally connected to both parties. G.729 Annex-B and G.723.1
Annex-A include an integrated VAD function, but otherwise performs the same as G.729 and G.723.1, respectively.

In Cisco CallManager, VAD can be enabled (it is disabled by default) with these service parameters:

SilenceSuppressionSystemWide—This parameter selects the VAD setting for all skinny endpoints (for example:
Cisco IP Phones and Skinny gateways)

SilenceSuppressionWithGateways—This parameter selects the VAD setting for all MGCP gateways. This does not
have an effect on H.323 gateways. VAD on H.323 gateways must be disabled on the gateway.

You can find these service parameters under Cisco CallManager Administration (Service > Service Parameters >
select_server > Cisco CallManager).

RTP Header-Compression or Compressed RTP (cRTP)

All VoIP packets are made up of two components: voice samples and IP/UDP/RTP headers. Although the voice samples are
compressed by the Digital Signal Processor (DSP) and can vary in size based on the codec used, these headers are a constant
40 bytes in length. When compared to the 20 bytes of voice samples in a default G.729 call, these headers make up a
considerable amount of overhead. With cRTP, these headers can be compressed to two or four bytes. This compression offers
significant VoIP bandwidth savings. For example, a default G.729 VoIP call consumes 24 Kb without cRTP, but only 12 Kb
with cRTP enabled.

Because cRTP compresses VoIP calls on a link-by-link basis, both ends of the IP link need to be configured for cRTP.

In Cisco IOS Software Releases 12.0.5T and earlier, cRTP is process-switched, severely limiting the scalability of cRTP
solutions due to CPU performance. Most of these issues have been resolved through various cRTP performance
improvements introduced in Cisco IOS Software Releases 12.0.7T through 12.1.2T. This is a summary of the history.

cRTP is process-switched in Cisco IOS Software Release 12.0.5T and earlier.

In Cisco IOS Software Release 12.0.7T, and continuing in 12.1.1T, fast-switching and Cisco Express Forwarding-
switching support for cRTP is introduced.

In Cisco IOS Software Release 12.1.2T, algorithmic performance improvements are introduced.

Moving cRTP into the fast-switching path significantly increases the number of RTP sessions (VoIP calls) that VoIP
gateways and intermediate routers can process.

Heuristics for Compression

As RTP does not have a distinct packet header of its own, an RTP stream (for cRTP) is distinguished from a UDP stream
(cUDP) by using heuristics. The exact heuristics used at present in order to detect RTP packets for compression are:

The destination port number is even.

The destination port number is in the range 16384-32767 or 49152-65535.

The RTP version field is set to two.

The RTP extension field is set to zero.

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