Bandwidth Requirements


The bandwidth requirements for voice traffic depend on many factors, including the number of simultaneous voice calls, grade of service required, codec and compression techniques used, signaling protocol used, and network topology.

On a WAN, for example, the bandwidth required for all applications, including voice, should be no more than 75 percent of the available bandwidth on the link;[7] the rest is for overhead, including routing protocol traffic.

The following sections detail how to calculate the WAN bandwidth that is required to support a number of voice calls with a given probability that the call will go through.

Definitions

To determine how much bandwidth is required, you must understand the following terminology:

  • Grade of service (GoS)

  • Erlangs

  • Centum call seconds (CCSs)

  • Busy-hour traffic (BHT)

Key Point

The grade of service is the probability that a call will be blockedthe caller will get a busy signal because of insufficient capacityduring the busiest hour.


The GoS is written as Pxx, where xx is the percentage of calls that are blocked. For example, a GoS of P05 means that a 5 percent probability exists that callers will be blocked when they call during the busiest hour.

Erlangs and CCSs are common methods of measuring voice traffic.

Key Point

One Erlang equals 1 hour, or 3600 seconds, of a telephone conversation.

A CCS is 1/36 of an Erlang. One Erlang therefore equals 36 CCSs.

The BHT, in Erlangs or CCSs, is the number of hours of traffic during the busiest hour of operation. BHTs are calculated as follows:

• To calculate the BHT in CCSs, multiply the number of calls in the busiest hour by their average duration in seconds, and divide the result by 100.

To calculate the BHT in Erlangs, multiply the number of calls in the busiest hour by their average duration in seconds, and divide the result by 3600.


For example, 1 hour of conversationone Erlangcan be one 60-minute call, three 20-minute calls, or fifteen 4-minute calls. Receiving 200 calls with an average length of 3 minutes in the busiest hour is 600 minutes or 36,000 seconds of traffic; this would be 10 Erlangs, or 360 centum call seconds of traffic.

Erlang tables show the potential traffic for a specified number of circuits at a given probability of receiving a busy signal. The potential traffic in the Erlang tables is the BHTthe number of hours of traffic during the busiest hour of the telephone system's operation.

One place that Erlang tables and calculators can be found is at http://www.erlang.com; other sites can also be found by using your favorite search engine. An Erlang B table is the most common traffic model that determines the number of circuits required for a given amount of busy-hour traffic and a required grade of service. For example, according to the Erlang B calculator, a BHT of 4.46 Erlangs at a GoS of P01 requires ten circuits.

After the number of circuits required is determined, the bandwidth required on the WAN for voice calls can be determined. This required bandwidth is also known as the trunk capacity.

Calculating Trunk Capacity or Bandwidth

Key Point

The trunk capacity for voice calls can be calculated by the following formula:

Trunk capacity = Number of simultaneous calls to be supported * Bandwidth required per call


The first component of this formula, the number of simultaneous calls to be supported, is simply the number of circuits required for the known amount of traffic, as calculated from the Erlang tables.

Note

If 100 percent of calls must go through, Erlang tables are not required; instead, the maximum number of simultaneous calls required should be used.


The bandwidth required for one call depends on the codec used and whether cRTP and VAD are used.

Using IP/UDP/RTP headers of 40 bytes and assuming that the Point-to-Point Protocol (PPP) is used at Layer 2 (so the Layer 2 header is 6 bytes), the following calculations can be made:[8]

Voice packet size = Layer 2 header size + IP/UDP/RTP header size + Voice payload size

Voice packets per second (pps) = Codec bit rate / Voice payload size

Bandwidth per call = Voice packet size (bits) * Voice pps

For example, the bandwidth required for a G.729 call (8-kbps codec bit rate) with a default 20-byte voice payload is calculated as follows:

Voice packet size (bytes) = Layer 2 header (6 bytes) + IP/UDP/RTP header (40 bytes) + Voice payload (20 bytes) = 66 bytes

Voice packet size (bits) = 66 bytes * 8 bits per byte = 528 bits

Voice pps = 8-kbps codec bit rate / (8 bits per byte * 20-byte voice payload size) = 8-kbps codec bit rate / 160 bits = 50 pps

Bandwidth per call = Voice packet size (528 bits) * 50 pps = 26.4 kbps

Table 7-3 summarizes the required bandwidth for G.711 and G.729 codec calls.

Table 7-3. Voice Bandwidth Requirements

Codec

Payload Size (bytes)

Bandwidth Required per Call (kbps)

G.711 (64-kbps)

160

82.4

G.729 (8-kbps)

20

26.4


VAD and cRTP reduces the bandwidth required per call. The results of the calculations for G.711 and G.729 at 50 pps are illustrated in Table 7-4.

Table 7-4. Per-Call Voice Bandwidth Requirements

Codec

Payload Size (bytes)

Bandwidth Required per Call Without cRTP or VAD (kbps)

Bandwidth Required per Call with cRTP (kbps)

Bandwidth Required per Call with cRTP and VAD (kbps)

G.711

160

82.4

67.2

33.6

G.729 (8-kbps)

20

26.4

11.2

5.6


CAUTION

Including VAD in bandwidth calculations can result in insufficient bandwidth being provisioned if the calls do not include as much silence as assumed and when features such as music on hold are used.


Recall that the trunk capacity for voice calls can be calculated by the following formula:

Trunk capacity = Number of simultaneous calls to be supported * Bandwidth required per call

As an example of calculating the trunk capacity, assume that G.729 compression is used over a PPP connection at 50 pps and cRTP is used. From Table 7-4, 11.2 kbps are used by each call. If five simultaneous calls are to be supported, 5 * 11.2 = 56 kbps are required for the voice calls. (The bandwidth for other traffic that is to be on the link must also be accounted for so that no more than 75 percent of the available bandwidth on the link is used.)

Signaling Traffic Bandwidth

Assuming that call control traffic must be sent to the CCM at a central site, the signaling traffic bandwidth requirements between the central site and remote sites depend on the number of IP phones and gateway devices.

In the case of a remote branch where no Telephony Application Programming Interface (TAPI) applications (such as Cisco IP Communicator software phone) are deployed, the recommended bandwidth needed for call control traffic can be determined from the following formula:[9]

Bandwidth (bps) = 150 * Number of IP phones and gateways in the remote site

In the case where a TAPI application is deployed at the remote site, the recommended bandwidth is higher because the TAPI protocol requires more messages to be exchanged between CCM and the endpoints. The formula then becomes as follows:[10]

Bandwidth with TAPI (bps) = 225 * Number of IP phones and gateways in the remote site

Note

The previous two formulas assume that an average of ten calls per hour is made on each phone. The referenced document provides more advanced formulas that should be used if this assumption is not true.





Campus Network Design Fundamentals
Campus Network Design Fundamentals
ISBN: 1587052229
EAN: 2147483647
Year: 2005
Pages: 156

flylib.com © 2008-2017.
If you may any questions please contact us: flylib@qtcs.net