# Calculating Bandwidth Requirements

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1 Calculating Bandwidth Requirements Codec Bandwidths This topic describes the bandwidth that each codec uses and illustrates its impact on total bandwidth. Bandwidth Implications of Codec 22 One of the most important factors for the network administrator to consider while building voice networks is proper capacity planning. Network administrators must understand how much bandwidth is used for each VoIP call. With a thorough understanding of VoIP bandwidth, the network administrator can apply capacity-planning tools. Following is a list of codecs and their associated bandwidth: G.711: The G.711 pulse code modulation (PCM) coding scheme uses the most bandwidth. It takes samples 8000 times per second, each of which is 8 bits in length, for a total of 64,000 bps. G.726: The G.726 adaptive differential pulse code modulation (ADPCM) coding schemes use somewhat less bandwidth. While each coding scheme takes samples 8000 times per second like PCM, it uses 4, 3, or 2 bits for each sample, thereby resulting in total bandwidths of 32,000, 24,000, or 16,000 bps. G.728: The G.728 low-delay code excited linear prediction (LDCELP) coding scheme compresses PCM samples using codebook technology. It uses a total bandwidth of 16,000 bps. G.729: The G.729 and G.729A Conjugate Structure Algebraic Code Excited Linear Prediction (CS-ACELP) coding scheme also compresses PCM using advanced codebook technology. It uses 8000 bps total bandwidth Cisco Networking Academy Program: v1.0 Copyright 2005, Cisco Systems, Inc.

2 G.723: The G.723 and G.723A multipulse maximum likelihood quantization (MPMLQ) coding schemes use a look-ahead algorithm. These compression schemes result in 6300 or 5300 bps. The network administrator should balance the need for voice quality against the cost of bandwidth in the network when choosing codecs. The higher the codec bandwidth, the higher the cost of each call across the network. Copyright 2005, Cisco Systems, Inc. Introduction to VoIP > Calculating Bandwidth Requirements 2-31

3 Impact of Voice Samples and Packet Size on Bandwidth This topic illustrates the effect of voice sample size on bandwidth. Impact of Voice Samples 23 Voice sample size is a variable that can affect total bandwidth used. A voice sample is defined as the digital output from a codec DSP that is encapsulated into a protocol data unit (PDU). Cisco uses DSPs that output samples based on digitization of 10 ms-worth of audio. Cisco voice equipment encapsulates 20 ms of audio in each PDU by default, regardless of the codec used. You can apply an optional configuration command to the dial peer to vary the number of samples encapsulated. When you encapsulate more samples per PDU, total bandwidth is reduced. However, encapsulating more samples per PDU comes at the risk of larger PDUs, which can cause variable delay and severe gaps if PDUs are dropped. Example: Encapsulated Bytes Calculation Using a simple formula, it is possible for you to determine the number of bytes encapsulated in a PDU based on the codec bandwidth and the sample size (20 ms is default): Bytes_per_Sample = (Sample_Size * Codec_Bandwidth) / 8 If you apply G.711 numbers, the formula reveals the following: Bytes_per_Sample = (.020 * 64000) / 8 Bytes_per_Sample = 160 The figure illustrates various codecs and sample sizes and the number of packets that are required for VoIP to transmit one second of audio. The larger the sample size, the larger the packet, and the fewer the encapsulated samples that have to be sent (which reduces bandwidth) Cisco Networking Academy Program: v1.0 Copyright 2005, Cisco Systems, Inc.

6 Specialized Encapsulations This topic describes considerations for specialized encapsulations for VoIP. Specialized Encapsulations X.25 over TCP/IP IPv6 over IPv4 L2F Others 26 There exist many other encapsulations to consider when transporting VoIP. Specialized encapsulations include protocol-specific encapsulation such as X.25, experimental encapsulations such as IPv6 over IPv4, Layer 2 Forwarding (L2F) Protocol, and other vendorspecific encapsulations. Each must be considered when calculating total bandwidth. Copyright 2005, Cisco Systems, Inc. Introduction to VoIP > Calculating Bandwidth Requirements 2-35

7 Calculating the Total Bandwidth for a VoIP Call This topic calculates the total bandwidth required for a VoIP call using codec, data link, and sample size. Total Bandwidth Required 27 Codec choice, data-link overhead, sample size, and compressed RTP have positive and negative impacts on total bandwidth. To perform the calculations, you must consider these contributing factors as part of the equation: More bandwidth required for the codec = more total bandwidth required More overhead associated with the data link = more total bandwidth required Larger sample size = less total bandwidth required Compressed RTP = significantly reduced total bandwidth required Example: Total Bandwidth Calculation The following calculation was used to produce the figure: Total_Bandwidth = ([Layer_2_Overhead + IP_UDP_RTP Overhead + Sample_Size] / Sample_Size) * Codec_Speed For example, assume a G.729 codec, 20-byte sample size, using Frame Relay without CRTP: Total_Bandwidth = ([ ]/20) * 8000 Total_Bandwidth = 26,400 bps 2-36 Cisco Networking Academy Program: v1.0 Copyright 2005, Cisco Systems, Inc.

9 VAD is enabled by default for all VoIP calls. VAD reduces the silence in VoIP conversations but it also provides comfort noise generation (CNG). Because you can mistake silence for a disconnected call, CNG provides locally generated white noise to make the call appear normally connected to both parties. Example: VAD Bandwidth Savings The figure shows examples of the VAD effect in a Frame Relay VoIP environment. In the example using G.711 with a 160-byte payload, the bandwidth required is 82,400 bps. By turning VAD on, you can reduce the bandwidth utilization to 53,560 bps. This is a bandwidth savings of 35 percent Cisco Networking Academy Program: v1.0 Copyright 2005, Cisco Systems, Inc.

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