Media Cards

There are two types of media cards that can be used to provide additional DSP resources to the MG1000E. The Media Card 32 (MC32) and the Media Card 32 Security (MC32S) both provide 32 channels of DSP Resources to the system. The difference between the two is that the MC32S provides the addition of Secure Real-Time Protocol (SRTP). SRTP encrypts the IP media path to and from the DSP channels on the MC32S card. It also offers improved echo performance over the MC32.

Media Gateway Controller

The Media Card Gateway controls the interface cards and application cards in the MG1000E Chassis. On a hardware level the MGC provides conference channels and tone generation for the phones. DSP Daughterboards are connected to the MGC to provide tone and conference functions between TDM and IP phones. The MGC is a replacement for the legacy Small Systems Controller (SSC). It has about 10 x the processing power of the SSC and four times the memory. It uses a standard Compact Flash for internal storage and has two internal ports for the installation of DSP Daughterboards. There are a total of six Ethernet ports on the MGC, four on the front (two ELAN, two TLAN), and two that are available through the backplane via a 50-pin to Serial/ELAN/TLAN Adapter.

There are two types of DSP Daughterboards that can be placed into the MGC. The 96-port DSP Daughterboard can be placed in Slot 2 only, and provides 96 channels of voice. The 32-port DSP Daughterboard can be placed in either Slot 1 or Slot 2. Using one or two DSP32 DBs you can achieve 32 or 64 channels of voice, and using a standalone DSP96 DB or a DSP32 and a DSP96 you can achieve 96 or 128 channels of voice. If the system requires more than 128 channels of voice to be converted from IP to TDM a separate Voice Gateway Media Controller card will be required. The DSP resources provide VoIP CODECs, compression, and echo cancellation.

The MGC provides 60 channels of Tone and Digit Switching (TDS), sixteen Digitone Receivers (DTR) or Extended Tone Detector (XTD) units, and additional tone service ports. It also provides two loops of 30 conference units each, for a total of 60 conference channels.

Installation and Commisioning a CS1000E with Release 5.5

There are a few options when setting up a CS1000E system as to the hardware that is used. A basic configuration will include one or two call servers, one or two signaling servers, a Media Gateway Controller, a data switch, and any trunking resources and voice media gateways.

In previous releases of the CS1000 series of hardware the signaling server and the Small Systems Controller (SSC) were co-located on one card. With the release of the 5.0 software this is no longer the case – now the signaling server is on a separate CP PM card, and the SSC is no longer used, instead we use a Media Gateway Controller. The Media Gateway Controller (MGC) also contains slots for two DSP Daughterboards which provide Digital Signal Processing resources, eliminating the need for a separate card for a Media Card. A separate media card may still be installed to increase the number of resources available.

I will now discuss the physical components of the Media Gateway 1000E.

Analog to Digital Conversion

When communications occur across VoIP the speech from one end is converted to a binary format and is transmitted across a data network to the receiving end. At the receiving end it is converted back to an analog voice signal. A signal may be converted multiple times, depending on the path it takes. The basic process of this conversion is as follows:
First, the audio signal is filtered to 300-3600 Hz, as most human communication takes place in this range. The filtered signal is then sampled at a rate of 8,000 Hz (8000 samples/second), to fit with the Nyquist-Shannon Sampling Theorem. This results in a Pulse Amplitude Modulation (PAM) signal. The PAM signal is then converted into a Pulse Code Modulation (PCM) signal, which is a digital signal. Since we only have a limited amount of space in the PCM signal (it is only 8 bits!) we must find a method of mapping the analog signal to this 8-bit PCM signal. There are two methods used to do this: μlaw and alaw. μlaw is used in North America and Japan, while alaw is used in the rest of the world. These standards assign more bits to represent the lower frequencies where most communication occurs, and fewer bits for the higher frequencies, which contain less communication. These PCM samples are then packaged into packets and shipped across the network.

There are two main components that perform this voice sampling and compression: Digital Signal Processors (DSPs) and CODECs. A Digital Signal Processor is a specialized multiprocessor that performs complex operations on digital signals. A CODEC is an algorithm that converts analog signals into digital outputs. The digital output can be transmitted over a data network and then reconverted to an analog signal at the other end. A CODEC can compresss and convert voice to digital data to minimize bandwidth requirements. A CODEC can compress a standard 64 Kbps voice signal down to as low as 5.38 Kbps. I will go into further detail on CODECs in our next section.