KHARKIV NATIONAL UNIVERSITY OF RADIO ELECTRONICS
MSGCS Laboratory
Work #2
Lines to Voice over IP Bandwidth Calculator
Purpose: Laboratory work is devoted to the investigation of the most commonly used coding algorithms for VoIP technology with the help of Lines to VoIP Bandwidth Calculator.
Basics of the Voice over IP
Voice over IP (VoIP) is most commonly transported as a digitally encoded stream using the Real-time Protocol (RTP) [RFC3550] over UDP; RTP is the transport layer protocol, which deals with the delivery of the VoIP bearer stream from sender to receiver. Signaling protocols such as the Session Initiation Protocol (SIP) [RFC3261] may be used to set up the RTP bearer streams and to determine the media formats (i.e. codecs) that will be used. The key factors that determine the impact that variations in networks SLA characteristics such as delay and loss have on VoIP are the codec that is used to encode the signal and the specific details of the end-system implementation. For example, some codecs may be less tolerant to loss than others, while a poor end-system implementation may be less tolerant to jitter.
techniques such as Algebraic Code Excited Linear Prediction (ACELP). ACELP breaks a sampled input signal into blocks of samples; these blocks or frames, which are typically 20 ms, are processed as whole units. In processing a frame, the encoder uses a technique called analysis-by-synthesis to determine which input parameters, when passed through a synthesizing filter, would result in reconstructed speech closest to the original speech signal. The encoder then uses a codebook to reference the inputs to the filter; the reference is sent to the decoder, which shares the same codebook, and which applies the respective inputs to the same synthesis filter to reconstruct the speech. There are a number of algorithms that have derived from ACELP, including Low-Delay Code Excited Linear Prediction (LD-CELP) and Conjugate Structure ACELP (CS-ACELP).
The codecs available for VoIP vary in complexity, in the bandwidth they need, and in the delivered call quality perceived by the end-user. Algorithms that are more complex may provide better perceived call quality, but may incur longer processing delays; Figure 1 shows the functional components in VoIP end-systems, which contribute to delay.
Figure 1 – VoIP end-systems components of delay
Here is a brief introduction to the calculator. For more detailed information, press the Help button. For a running record of the results calculated, press the Results
button. Both these functions open new browser windows on your desktop.
This calculator can be used to estimate the bandwidth required to transport a given number of voice paths through an IP based network. Reverse calculations are also possible. These estimate the number of voice paths that can be transmitted though an IP network if the available bandwidth is known.
Before a calculation can be performed, details of the voice compression scheme must be entered into the first two areas of the calculator.
1. Use the first drop down box to select the CODEC being used. CODECs convert analogue voice signals into data streams through sampling and quantization. CODECs vary in their quality and delay characteristics and G.723.1 and G729A are the most common CODECs used for Internet voice transmission.
quality. Lower durations require more bandwidth. However, if the duration is increased, the delay of the system increases, and it becomes more susceptible to packet loss; 20 ms is a typical figure.
To perform a calculation:
select the value which you want to calculate by clicking on the radio button representing that parameter;
enter the known value into the edit box representing the other parameter;
when the Calc. button is pressed, the result will be shown in the other edit box.
For example, to work out how much bandwidth is required to transmit 10 voice paths (or channels) through an IP network using the G.723.1 (6.4kbps) coding scheme with 30ms packet duration, follow these steps:
1. Use the Coding algorithm drop down list to select G.723.1 (MP-MLQ) 6.4 kbps compression.
2. 30ms should automatically be selected as the Packet duration. With this coding scheme, lower packet durations are not possible.
3. Ensure that the Unknown radio button in the Bandwidth area is selected. 4. Enter 10 into the edit box within the Voice paths area.
5. Press the Calc. button. After a short time, 171 should appear in the Bandwidth
edit box, indicating that a bandwidth if 171kbps would be required.
Task List:
1. Investigate different coding algorithms using Lines to VoIP Bandwidth Calculator with different packet duration.
2. Build the graphs of the dependencies IP Bandwidth of packet duration for every coding algorithm using MS Excel.
drop down list, and perform the Bandwidth calculation for every possible value of packet duration (number of Voice paths choose 10). Save results in MS Excel spreadsheet and plot the graph of the dependency IP Bandwidth of packet duration. Continue investigation for the rest of Coding algorithms.
Example:
Coding algorithm: G.711 (PCM) 64 kbps uncompressed.
Figure 3 – Graphical results
Useful links:
http://www.erlang.com/ http://www.voip-info.org/wiki/view/Codecs http://voip.about.com/od/voipbasics/a/voipcodecs.htm http://www.broadcom.com/support/broadvoice/codec_comparison.php 800 900 1000 1100 1200
10 20 30 40 50 60 70 80 90
B an d wi d th (k b p s)
Packet Duration (ms)
