Voice over IP Training - SIP and MGCP Set-Up DO-410521-TM-A

Training – Voice over IP
1. Hands-On - Setting up VoIP configurations in Lab
Physical VoIP Network:
Introduction
The two common VoIP configurations are SIP and MGCP. The sample set-up in the VoIP Demonstration Room can use both, having similar physical network properties.
a. SIP Proxy Server
The SIP Proxy Server currently in use is the SIP Exchange Router (SER) with Linux RedHat 9.0. We have also tested using SER with RedHat 7 and 8, and the MS Windows-based Brekeke OnDO SIP Server. Paragraph 8 of DO-406249-TC Issue 1 contains a detailed description of the configuration, but for convenience, they are produced here:
b. MGCP Proxy Server
We currently use the MediaMind MGCP Server. Paragraph 7 of DO-406249-TC Issue 1 refers. To configure, use the following commands produced here for your convenience.
2. Hands-On - Voice Quality Tests
Malden Digital Speech Level Analyser – Background Information
Voice quality is measured using a unit known as a ‘Malden Digital Speech Level Analyser’. Processes compare the original and received signal and determine Mean Opinion Score (MOS) predictions on a scale of 1-5. The linguistically motivated artificial speech signal contains a representative selection of sounds put together in legal strings to provide a brief speech signal that does not contain the redundancy of real speech. Note: Detailed hardware and software installation instructions can be found in the User Manual at /downloads.htm.
Table 1. MOS – Interpretation of Results
DSLA QuickStart Example
Start the DSLA software and select View Channel A and View Channel B. Ignore any error messages relating to hardware not being attached.
We will be using QuickStart American Telephone-Telephone files, so need to select the filepaths. Select View, PAMS, and navigate to the dsla\quick start\american\telephone-telephone.sch files for both inputs. Click OK.
We now need to create a schedule of events – the QuickStart program has provided us with the reference data as shown in the A to B and B to A events listed.
Note that Telephone A is going to call Telephone B. Telephone B will not return a call to A (although the traffic will pass in both directions).
From the Schedule module, select A calls B. Notice how some parameters appear in both Channel A and Channel B events boxes. These have been set to default settings, which we need to adjust slightly for our system to work.
Change the DTMF Sequence to read 7301 and change the Output level to 0.00 dBm. Select Apply and Close.
Ensure that:
Channel A is connected to one voice unit, and Channel B is connected to the other unit which is configured to respond to DTMF 7301.
Either earphones or speakers are connected to the DSLA unit.
Bring the Scheduler window into view again, and place the cursor over Telephone A - Telephone B in the list and single click the mouse. Now click the Play button.
You can observe the commencement of the test by listening to the transmitted reference data through the earphones or loudspeaker. The following dialogue box appears and shows the progress through the particular test element:
Following completion of each test a report is made to the user displaying the Listening Effort and Listening Quality results. If this is not visible, select View, PAMS.
You will also note that the Reference Surface (transmitted) waveforms and Degraded Surface (received) waveforms are visible in graphical form. The third graph represents the difference between the reference and Degraded Surface.。