FXO stands for Foreign Exchange Office. An FXO port terminates a POTS line from the central ‘office’. XOP Networks provides analog FXO based audio conference bridge. For a 16 port systems, the 16 POTS lines from the Central Office can be connected to the bridge. In this case each FXO port on the bridge appears as a telephone to the central office switch.

FXS stands for Foreign Exchange Station. You connect a regular analog phone or a ‘station’ to a FXS port. This port provides battery feed to the phone. XOP Networks Ringdown Firebar Conference Server provides FXS ports. In a typical deployment analog Red phones can be connected to it over copper wire pairs.

T1 refers to a Time Division Multiplex based carrier routinely used in North America. This interface carries 24 voice channels. Each voice channel uses 64 Kbits/sec. A T1 line also has an 8 Kbits/sec signaling/framing channel. Therefore the bit rate for a T1 carrier is 24 x 64K + 8000 = 1.544 Mb/s.

There are two types of framing formats used with T1 carriers. These are called D4 and ESF.

The T1 carrier needs to maintain a certain 1’s density so that the network equipment at each end can recover timing clock even if no traffic is being carried. There are two mechanisms in use for that purpose. These are referred to as AMI (Alternate Mark Inversion) or B8ZS (bipolar 8-zero substitution).

The older T1 trunks (> than 10 years in service) are generally set up for D4/AMI and those in last 5-10 years are generally set up for ESF/B8ZS.

The XOP Networks equipment supports both flavors of T1s.

A communication line that was developed by European standards that multiplexes thirty voice channels and two control channels onto a single communication line. The E1 line uses 256 bit frames and transmitted at 2.048 Mbps.

E1 refers to a TDM (Time Division Multiplex) based carrier used in Europe and South America. Japan uses a variant called J1.

SIP Trunk is a telephony term for a Data path that serves the function of a traditional T1 or E1 trunk but uses the SIP (Session Initiated Protocol). This facilitates the integration of voice and data on the same physical carrier.

Before SiP became ubiquitous, some early Voice over Data networks were set up using the H.323 protocol.

H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that may not provide a guaranteed Quality of Service (QoS). H.323 specifies techniques for compressing and transmitting real-time voice, video, and data between a pair of videoconferencing workstations. It also describes signaling protocols for managing audio and video streams, as well as procedures for breaking data into packets and synchronizing transmissions across communications channels.

E1’s can be deployed in a 75 ohm architecture where the signal traveled on Coax cable and used BNC connectors. They can be deployed in a 120 ohm architecture where the signal travels over twisted pair and uses the more familiar RJ-45 connector. A Balun is a device that allows a 75 Ohm E1 to be converted to a 120 ohm E1 (or vise versa) to facilitate connection to equipment that does not support the native connector.

CAS (Channel Associated Signaling) is a protocol used over T1 and E1 trunks where each DS0 or channel has bits that carry the call control signaling states, ie idle, seize, wink, release. This is inefficient from a network standpoint because the entire DS0 is used to convey just a few bits of signaling before the voice path is established.

ANI (Automatic Number Identification) is the field in a CAS call that carries the Caller ID information. In PRI this field is called the Calling Number, In a SiP invite, this is the FROM address. This information is critical for displaying Caller ID at the end device.

DNIS (Dialed Number Identification Service) is the field in a CAS call that carries the number dialed commonly called the destination. In PRI this field is the Called Number. In a SIP invite, this is the TO address. This number is used by PBX’s, XOP USN’s, and a variety of other call handling systems to select the menu choices, call routing, and customer service representative information display based on the incoming telephone number.

Service selection is the term for using either the called number or calling number, or the combination of the two to automatically route calls and launch services. On the XOP USN we can use the ANI or the DNIS or the combination of a specific ANI and DNIS to automatically launch services, ie a call from your cellphone to the USN will automatically put you into your conference as a moderator. It is an excellent way to make it easier for users to access the system because it removes the need to remember and enter a pin code.

The terms Network and User are designed to refer to the Carrier or Service Provider which is the Network, and the CPE (Customer Premises Equipment)

XOP Equipment can support Non facility Associated Signaling (NFAS) if a customer requires it. By default this capability is turned off to keep the network operation simple. This capability allows a single D channel to be used across multiple T1 based ISDN PRI lines. This capability is rarely used on E1 based PRI lines.

A typical configuration for a SIP trunk on XOP Networks equipment is: G.711 codec (Alaw or MuLaw), RFC 2833 for DTMF relay.

On TDM phone circuits the DTMF tones are used to enter dialed digits, PINs, card numbers etc. These DTMF tones are carried inband on the TDM network. These functions are also needed in VoIP network. When compressed voice is used on the VoIP networks, e.g., with G.729a/b or G.723 codecs, DTMF tones get distorted if carried inband. In such cases DTMF tones are carried as data in special packets. The circuitry at the receive end simply reads to the packet to get to the digit pressed. This mechanism of using special packets for carrying DTMF digits is described in RFC 2833. For VoIP networks that uses g.711 codec, it is not necessary to use RFC 2833 as the DTMF tones do not get distorted in that case. The codec used and whether RFC 2833 is used or not is carried with in the SDP part of the SIP INVITE message. Both transmit and receive end agree with the protocol to be used for DTMF relay as part of the initial signaling handshake.

Linux is a open source, much more inherently stable and secure operating system. Being open source it keeps evolving based on the work effort of thousands of developers worldwide. Its roots are based in the UNIX operating system. Several companies re-package the open source software and then market it under their brand names.

Yes.

XOP Networks’ Release 4.x uses CentOS 5.0. Its current software release 5.2 uses CentOS 5.4. We will continue to adopt newer versions of the operating systems going forward based on their compatibility with hardware and features available in a given Linux distribution.

Red Hat is a name of a company that provides Linux distributions. It also provides support for them. For more details please visit its website.

It is the name of a LINUX distribution. XOP Networks products with Release 5.0 or higher use CentOS Version 5.4.

The computing power of CPUs used in PCs and Servers continues to grow. Basic idea behind Host Media Processing is to use the core CPU of a computer to do Digital Signal Processing functions (e.g., echo cancellation, AGC, conferencing etc.) as well in addition to its normal house keeping tasks. Until such powerful CPUs were available, the DSP algorithms used to run on dedicated DSP hardware modules. By terminating a VoIP based SIP trunk on a server that is also running HMP based conferencing software, we can provide conferencing equipment that has no dedicated DSP hardware. Using this approach XOP Networks ships a 1U server that can provide conferencing and other value added services applications for several hundred VoIP ports.

Yes. We provide systems that have network facing T1/E1 trunk modules and Ethernet port for connectivity to the SIP trunk. The Host Media Processing software runs on the core CPU of the server. This software is provisioned based on the number of total voice ports needed. For example, we can provide a 50 port conference bridge that has a T1 based network module that provides connectivity to 24 TDM ports and has a SIP trunk with 26 channels for connectivity to VoIP network. In this scenario XOP conferencing application will be able to place upto 50 people in a conference room -24 entering from the TDM side and 26 entering from the VoIP side.

RAID (Redundant Array of Independent Disks), a system of multiple hard drives for sharing or replicating data. XOP Networks equipment supports disk mirroring between two hard drives using RAID1.

A RAID 1 creates an exact copy (or mirror) of a set of data on two or more disks. This is useful when read performance or reliability are more important than data storage capacity. Such an array can only be as big as the smallest member disk. A classic RAID 1 mirrored pair contains two disks, which increases reliability geometrically over a single disk. Since each member contains a complete copy of the data, and can be addressed independently, ordinary wear-and-tear reliability is raised by the power of the number of self-contained copies. (from Wikipedia).

NEBS (Network Equipment-Building System) describes the environment of a typical United States RBOC Central Office. NEBS is the most common set of safety, spatial and environmental design guidelines applied to telecommunications equipment in the United States. (from Wikipedia).

DCB refers to XOP’s audio + web conference bridge. The Universal Service Node is a super set of the DCB in the sense that provides audio+web conferencing plus several other Services such as Visual Voicemail, Mass Notification, Firebar etc.

Digital Signal Processing is the science of manipulating digital samples of an analog signal. For example, for audio conferencing, digital samples from different participants are digitally added to create a combined signal. The digitally combined signal is converted back to analog by the end device and fed to the speaker of each participant.

Echo gets generated on voicecalls when there is a impedance mismatch at the 2 wire to 4 wire converter. Due to the mismatch part of the signal gets reflected back to the originator. If at the transmit end the reflected signal can be removed then echo will not be heard. The process of removing this unwanted echo is called echo cancellation. This is done using digital signal processing techniques.