Sangoma Enters Analog Telephony Market with Release of Remora A200 Analog Cards

March 9, 2006 by Garrett Smith

Sangoma has recently announced their A200/REMORA FXO/FXS Analog Telephony support System. It is now available at

The A200 and REMORA system together comprise the FXO/FXS version of Sangoma’s range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of analog voice traffic.

The A200 and REMORA cards support up to a total of 24 FXO and/or FXS connections. A single PCI slot host connection for all ports ensures common synchronous clocking for all channels. The base AFT architecture is shared with Sangoma’s A101, A200 and A104 cards ensuring proven 3.3v/5v, high performance PCI compatibility.

Like all the Sangoma AFT Series, the A200 and REMORA system is field upgradeable to take advantage of the hardware and software improvements as they become available. Also, the A201 supports Sangoma’s echo cancellation and voice enhancement board so that hardware voice enhancement is available for analog as well as TDM voice.


The A200 consists of a REMORA daughterboard mounted on the AFT PCI card . The REMORA card has two sockets each of which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS ports respectively.

Up to four additional REMORA daughterboards can be mounted in empty slot positions beside the A201 assembly, connected to the A201 by a backplane bus connector.

Technical Specifications

  • Support for the Asterisk™ , Yate™ , OPAL™ PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications.
  • Single synchronous PCI interface for all 24 FXO/FXS ports.
  • Four RJ11 4 ports per REMORA card.
  • Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis.
  • Short 2U compatible mounting clips included for installation in 2U rackmount servers.
  • 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention.
  • Autosense compatibility with 5v and 3.3v PCI busses.
  • Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper interrupt sharing.
  • Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features related to voice and/or data can be added when they become available.
  • Power: 800mA peak, operational 300mA max at +3.3v or 5v.
  • Temperature range: 0 – 50C.
  • Optional DSP card on the A200d
  • G.168-2002 echo cancellation in hardware
  • 1024 taps/128ms tail per channel on all channel densities
  • DTMF decoding and tone recognition
  • Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction.

Wiring Connections

The A200 and Remora cards incorporate four, 4 pin RJ11 narrow jacks such as used in telephone handsets. Each A200/Remora is shipped with four 2M cables terminating in a narrow RJ11/4 plug at one end and a telephone-standard RJ11/6 plug at the other.

For those who need to hard wire the A200 system, Sangoma has a kit available consisting of 24 RJ11/4 plugs and a crimping tool.

(1) Sangoma Website, A200 / Remora Datasheet, 2005, (Nov 2005).

More from: Asterisk Garrett Smith

Hardware Echo Cancellation – Sangoma and Digium

In recent months, both Digium and Sangoma have released T1/E1 Interface cards for use with Asterisk Open Source PBX that provide hardware Echo Cancellation.

Echo can negatively impact QOS (Quality of Service) on a VOIP call and lead to an unsatisfactory user experience. There are two sources of echo on voice communication networks. The first, and most common cause of echo is impedance mismatches anywhere in the circuit-switched phone network. When telco cabling is spliced or terminated in connectors, or in the conversion of 4-wire phone circuits to 2 wires, a discontinuity occurs that causes an impedance mismatch on the phone circuit. The greater the extent of impedance mismatch, the more leakage of transmit audio on the receive side of the line when it terminates on a 2-wire phone circuit or device. The main cause of this impedance mismatch is the 2 to 4 wire hybrid that converts a 4-wire audio path to a local 2-wire loop. This tip-and-ring loop is the most-used type of telephone connection made to subscribers from your telephone company central office, and directly drives common analog telephone devices.

Acoustic coupling between the microphone and speaker of a telephone device leads to Acoustic echo, the second most common cause of echo on voice communication networks. Received caller audio can often “leak” to the microphone in cheaper speakerphones and hands-free cellular phones, because of sound pressure from a speaker or earpiece device

This inherent echo led to the development of Echo Cancellers, in order to improve the quality of voice communications. Echo cancellers are very complex digital signal processing devices, and the algorithms that drive them are produced by a handful of technically proficient companies. Naturally, as telecommunications evolves from the legacy PSTN network to the VOIP network, echo cancellation is still a very necessary component for ensuring QOS.

Digium offers hardware echo cancellation on their TE406P and TE411P telephony boards, as well as their new TDM2400 series full length analog PCI cards. Digium’s hardware echo cancellation provides 64ms across 32 channels; however, when it scales over 32 channels it is reduce to 16ms per channel across all channels.

Sangoma offers hardware echo cancellation on their A104D Quad T1 card, as well as on their as yet unreleased series of analog telephony boards. Unlike Digium where available ms of echo cancellation decreases as active channel density increases, Sangoma hardware echo cancellation provides a full 128ms of echo cancellation on all channels, regardless of density.

By providing hardware echo cancellation which extends to a full 128ms on all channels at full load, Sangoma provides a more robust compensation for echo which is more effective under extreme conditions. The echo tail length represents how long the effect of an echo extends after the time of the echo source. If a ping is transmitted at time zero, 16ms (16 ms = 128 taps at 8 samples per millisecond) of tail will deal with any echo from that ping that occurs within 16 ms. The echo itself dies away in about 8ms, so as long as the delay between the ping and the start of the echo is less than 8ms, then the echo canceller will work well. The problem arises when the signals get delayed. For instance traveling through one Telco switch adds about 5ms of two-way delay, so it is easy to see how a bit of extra switching could delay the echo right out of an echo canceller’s

The above image illustrates the echo from a “ping” on a real line where there are
switching delays. In cases of extreme echo as pictured above, normal 128 tap echo cancellation would be ineffective at cancelling the echo present on the line.

Software echo cancellation works quite well for electrical echoes that are within the 16ms/128 tap range. Where hardware echo cancellation really has benefit is where the delays are longer. The Sangoma echo canceller is a carrier-grade device with 1024 taps (128ms) of tail to handle the most severe echo problems.

More from: Asterisk Garrett Smith

FCC Deadline for VOIP Provider E-911 Compliance Passes

Providers of Voice over IP (VOIP) services may be restricted from enrolling new customers after the FCC specified date for E-911 compliance is past.

The FCC had previously imposed a 120 day window for VOIP providers to deploy reliable, enhanced 911 services in all their service areas by Monday, November 28, 2005.

Since the deadline passed, little public disclosure of E-911 compliance has been offered by the FCC or the affected VOIP providers. It had been previously estimated by the VON Coalition, a VOIP industry consortium, that about two-thirds of VOIP users would have access to E-911 services by the deadline.

Linksys Announces WBP54G 802.11G WIFI “Dongle”

Linksys has a cool new product on the market that should appeal to owners of the popular Linksys/Sipura line of Analog Adapters and SIP phones. The WBP54G adds 802.11G Wireless capabilities to any compatible Linksys/Sipura device, eliminating the need to run RJ45 network cable to network such devices.

Now you can locate a Linksys Phone Adapter or SIP Phone almost anywhere, without the cost and hassle of running network cables. The WBP54G was specially designed to convert a SIP phone or adapter into a wireless device, so it can connect to your home network without an Ethernet cable. This lets you put your phone where it’s most convenient for you, and not be constrained to the area around your Internet connection.

“To make installation even more convenient, the Wireless-G Bridge shares electrical power with the Phone Adapter, so only one AC Adapter (and wall plug) are needed. To get connected, just plug your existing Phone Adapter’s power jack to the Wireless-G Bridge, and the Bridge’s attached power and data cables to the Phone adapter. The included Setup Wizard makes it easy to configure the Bridge to your wireless network’s settings. To protect your data and privacy, all wireless transmissions can be encrypted with WEP or industrial-strength Wi-Fi Protected Access (WPA/WPA2) security.

New SIP PBX Appliance from D-Link – DVX-1000

D-Link has developed a SIP based PBX appliance called the DVX-1000

The DVX-1000 fulfills many of the phone system requirements of the SMB market, including popular features such as call forwarding, call hold, find me-follow me, and voicemail. Inbound calls are routed through the integrated auto-attendant and hunt groups to assist callers to their destinations. The DVX-1000 utilizes standard phone lines via an external phone line gateway or cost effective Internet Telephony

One DVX-1000 VoIP PBX appliance can support up to 25 extensions, which can be located in any physical/geographic location provided that location has Internet access. Multiple units can be peered to increase the number of extensions or unite a company that has many locations under a single PBX system. Additional extensions are added via license codes that are obtained via your reseller or directly from D-Link.

The DVX-1000 IP PBX is fully user configurable via a user friendly web configuration tool. The administrator assigns each extension a profile of telephony features, which allows the best match for a users job function. Each user can fine-tune their assigned profile via the web to match their daily business schedule.

In the world of traditional PSTN telecom, conferencing is typically an expensive external hardware or service. The DVX-1000 includes a phone conferencing bridge, which eliminates the need for an external conferencing server or service provider, and adds distinct value to the D-Link offering. Users are able to schedule and invite parties to conferences via a simple web configuration tool. Conference Notifications are sent out by e-mail, which includes the conference phone number and access codes.

The DVX-1000 uses advanced security features to protect your voice network from unauthorized access. To prevent hackers from breaching the system, the DVX-1000 uses MD5 SIP authentication encryption encoder software. The DVX-1000 also includes an integrated firewall for intrusion detection and protection against denial of service attacks.

The DVX-1000 features a fanless solid-state design offering years of non-stop operation. The compact housing can be easily fastened to the wall of your distribution closet or stacked with your existing Ethernet switches or PSTN Gateways. The DVX-1000 is designed with dual processors for supporting up to 25 simultaneous calls. Its class leading performance allows a 1-to-1 extension to phone line mapping, allowing it to scale with your business.

Digium Ships New TDM2400 Series Full Length Analog PCI Cards

Digium has begun shipping their new TDM2400 Series Full Length Analog PCI Cards to VoIP Supply.

The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium’s Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system.

Starting with a base PCB Board that is available with or without Echo Cancellation, the TDM2400 can be configured with FXS or FXO Resources, added in 4 port increments, up to a maximum of 24 total ports. This allows for a wide variety of FXS/FXO combinations, and in many cases eliminates the need for an external gateway or channel bank altogether.

The Digium TDM2400 Series boards are designed for use with Asterisk Open Source PBX

More from: Asterisk Garrett Smith

UTStarcom F3000 – First 2nd Generation WIFI Phone?

UTStarcom has been one of several successful entrants into the WIFI VoIP handset space in recent months. Their initial product, the F1000, set the preliminary standard in terms of usability, feature set and price point. Other vendors including Hitachi-Cable and Zyxel have also brought wireless SIP handsets to market based on the 802.11XX WIFI standard.

The initial crop of wireless VOIP phones all seemed to have their own individual strengths and weaknesses. With the upcoming release of the F3000, UTStarcom seems to have built upon the successful foundation established with the F1000, and have set about addressing some of the limitations and inadequacies of first generation WIFI VOIP handsets.

The UTStarcom F3000 has added support for the current encumbent 802.11G Wireless standard.

The UTstarcom F3000 promises increased security with the addition of WPA encryption, which utilizes the temporal key integrity protocol (TKIP). TKIP utilizes a hashing algorithm to scramble the keys and, by adding an integrity-checking feature, ensures that the keys have not been tampered with.

User authentication, which is generally missing in WEP, through the extensible authentication protocol (EAP). WEP regulates access to a wireless network based on a computers hardware-specific MAC address, which is relatively simple to be sniffed out and stolen. EAP is built on a more secure public-key encryption system to ensure that only authorized network users can access the network.

The preliminary spec we have seen for the UTStarcom F3000 promises to address one of the major factors which has limited the mass-adoption potential of first generation Wireless VoIP handsets, namely Handover/Roaming between different AP and SSID. The ability to “roam” between various access points and SSID within a WIFI network, without losing call connectivity of SIP registration in the process, will seriously bolster the case for WIFI VOIP adoption.

Finally, some aesthetic enhancements are evident in the new F3000, including a larger, color LCD screen and a clamshell form factor typical of current mobile/cellular/GSM phones.

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