Analog endpoint bridge
Connect conventional telephones, G3 fax machines, modems and legacy analog interfaces through two FXS ports.
The Mediatrix 4102S is a secure VoIP Analog Adaptor that interconnects analog telephones, faxes and modems into SIP-based systems. This page consolidates the key technical information directly into the HTML: interfaces, voice processing, SIP, fax, QoS, security, management, networking, LEDs, power, physical dimensions and installation requirements.
The 4102S links analog connections to IP networks and delivers the feature set required for secure VoIP deployments, legacy PBX integration, cloud telephony enablement and mass-managed branch office rollouts.
Connect conventional telephones, G3 fax machines, modems and legacy analog interfaces through two FXS ports.
Support for G.711 A-law/µ-law, G.726, G.729a/b, echo cancellation, DTMF, de-jitter buffer and packet loss concealment.
SIP 2.0 with UDP, TCP and TLS transport, encrypted signaling, SRTP media encryption and certificate-based security.
IPv4/IPv6, VLANs, QoS tagging, NAT, firewall, routing, PPPoE, DHCP client/server and LLDP-MED.
Zero-touch provisioning, web GUI, SSH, SNMP, TR-069/TR-104/TR-111 and event notifications for operator-scale management.
Alarms, traps, CDR, eMOS/RTCP-XR reporting, media quality statistics, PCM capture, packet capture and diagnostic traces.
This section is structured for buyers, engineers, integrators and search engines. It exposes the technical data in readable HTML tables instead of hiding the details in downloadable files.
| Analog ports | 2 x RJ-11 FXS connectors for conventional telephones or G3 fax machines. |
|---|---|
| Ethernet ports | 2 x 10/100 Base-T Ethernet RJ-45 connectors: LAN and WAN/uplink. |
| Power connector | External 12 Vdc, 1.5 A power supply input. |
| Reset / Default | Hardware reset/default switch to restore the unit to known values. |
| SIP | SIP 2.0, RFC 3261 and 3GPP alignment. |
|---|---|
| Transport | UDP, TCP and TLS. |
| Media | RTP/RTCP, RFC 3550. |
| Session description | SDP, RFC 4566. |
| DTMF | In-band, out-of-band RFC 2833 and SIP INFO. |
| Redundancy | DNS SRV redundancy support and multiple trunk support. |
| IP stack | IPv4 and IPv6 dual stack. |
| Voice codecs | G.711 A-law and µ-law, G.726, G.729a/b. |
|---|---|
| Voice channels | High-performance processing of up to 4 voice channels. |
| Echo cancellation | G.168 echo cancellation. |
| DTMF handling | DTMF detection and generation. |
| Silence handling | Silence detection/suppression and comfort noise. |
| Jitter | Configurable de-jitter buffer and packet length. |
| Packet loss | Packet loss concealment. |
| T.38 fax | Real-time T.38 fax over IP. |
|---|---|
| G.711 fax | Clear-channel G.711 fax and modem pass-through. |
| Legacy data | Designed to preserve analog data service transport over IP networks. |
| Use cases | Fax machines, modems, G3 fax terminals, legacy PBX analog endpoints and cloud telephony migrations. |
| Call features | Call forward, call transfer, call waiting, call hold and 3-way conference call. |
|---|---|
| Caller ID | On-hook and off-hook Caller ID. |
| Tones | Country tone presets, customizable tones and ring patterns. |
| Ringing | Distinctive ringing support. |
| MWI | Message Waiting Indication. |
| MoH | Music on Hold. |
| Signaling | Answer and disconnect signaling, battery reversal for pay phones. |
| Traffic control | Bandwidth limitation and traffic shaping. |
|---|---|
| IP QoS | TOS / DiffServ. |
| Layer 2 QoS | IEEE 802.1p/Q support. |
| Tagging | IEEE 802.1q and DSCP QoS tagging for media, signaling and management. |
| DoS protection | Denial of Service protection. |
|---|---|
| Signaling encryption | SIP over TLS. |
| Media encryption | SRTP with AES cipher, 128-bit. |
| Key management | MIKEY RFC 3830 / RFC 4567 and SDES RFC 4568. |
| Management security | TLS-encrypted configuration and management. |
| Certificates | X.509 certificate management and OCSP revocation status verification. |
| TLS version | TLS 1.3 with secure TLS ciphers such as ECDHE with AES-256 and SHA-384. |
| IP | IPv4 and IPv6. |
|---|---|
| Addressing | Multiple IP addresses per link or VLAN. |
| VLAN | Multiple VLANs per link. |
| DHCP | DHCP client and DHCP server. |
| WAN access | PPPoE, RFC 2516. |
| Authentication | IEEE 802.1x wired authentication. |
| Discovery | LLDP-MED, ANSI/TIA-1057. |
| Firewall | Stateful inspection, rate limitation and automatic blacklisting. |
| Routing | Static routing and NAPT. |
| Provisioning | Zero-touch provisioning and auto-provisioning via FTP, TFTP, HTTP and HTTPS. |
|---|---|
| Management protocols | TR-069, TR-104 and TR-111. |
| Interfaces | Web GUI, SSH and Telnet. |
| SNMP | SNMP v1, v2c and v3. |
| Access rights | Multiple levels of management access rights. |
| Notifications | Event notifications via Syslog, SIP, log file and SNMP traps. |
| Licensing | Remote activation of service licenses. |
| Factory options | Customizable factory settings. |
| Local switching | Local call switching for controlled routing scenarios. |
|---|---|
| Filtering | Call filtering and blocking. |
| Number manipulation | Calling/called number manipulation using regular expressions. |
| Routing criteria | Interface, calling/called party number, calling/called URI, time of day, day of week, date and other conditions. |
| SIP mapping | Mapping and transformation of call properties to and from SIP headers. |
| Hunt groups | Hunt group support. |
| Alarms | Alarms and traps. |
|---|---|
| Call quality | Call quality reporting using eMOS and RTCP-XR as per RFC 6035. |
| CDR | Call Detail Record support. |
| Statistics | Media quality statistics, CPU and memory usage. |
| Capture | PCM capture and IP network capture. |
| Traces | Diagnostic traces for operational troubleshooting. |
| Ready | Green steady ON when all lines are enabled; OFF when all lines are disabled; blinking 1 cycle per 4 seconds when at least one line is enabled and at least one line is disabled. |
|---|---|
| In-Use | OFF when lines are idle and unlocked; steady ON when lines are in use and unlocked; steady yellow during shutdown; blinking yellow at 1 cycle per second when locked. |
| ETH | Green blinking at variable rate when network traffic is present; green steady ON when connected with no traffic; OFF when not connected. |
| Power | Amber steady ON when restart is completed; OFF when the unit is not connected; blinking 1 Hz during restart. |
| External AC supply | 100-240 VAC external power supply. |
|---|---|
| Power input | External 12 Vdc, 1.5 A adapter. |
| Frequency | 50 Hz / 60 Hz. |
| Operating temperature | 0ºC to 45ºC according to datasheet; installation guide location requirement lists 0ºC to 40ºC. |
| Storage temperature | -20ºC to 70ºC. |
| Humidity | Up to 85%, non-condensing. |
| Datasheet dimensions | Height 3.1 cm, width 12.7 cm, depth 9.9 cm, weight 170 g. |
|---|---|
| Installation guide chassis data | Height 4.9 cm, width 22 cm, depth 17.6 cm, weight 458 g. |
| Installation note | Install on a flat surface, wall or equipment location with adequate ventilation and access for maintenance. |
| Clearance | Maintain at least 25 mm / 1 in. clearance in front, back, top and sides. |
| Operators | Connect legacy equipment in PSTN/TDM replacement projects, provide SIP termination for cloud telephony services and convert analog signaling to SIP for hosted UC and IP-Centrex. |
|---|---|
| System integrators | Integrate Unified Communications with legacy systems, keep existing telephony equipment in SIP migrations and interconnect branch offices to headquarters. |
| Legacy PBX | FXS ports, local call switching and user-defined call properties support integration into legacy PBX environments. |
| Bandwidth-sensitive sites | High-compression codec support helps conserve bandwidth in distributed or constrained environments. |
| Branch / SMB | Designed to shorten deployment time and enable secure cloud telephony services into branch offices and SMBs. |
The 4102S sits at the analog edge and converts existing voice, fax or modem endpoints into managed SIP-based communication.
This page centralizes the technical details normally searched by network architects, VoIP engineers, telecom integrators and purchasing teams: analog interfaces, LAN/WAN connectivity, SIP behavior, fax over IP, voice codecs, QoS, IPv4/IPv6 networking, VLAN support, NAT traversal, firewall functions, secure signaling, encrypted media, certificate handling, remote management, provisioning and physical installation constraints.
All core specifications are written directly in the HTML so Google can index the technical content without requiring a separate PDF datasheet.
Technical data is grouped by interface, voice, fax, network, security, management, physical and deployment categories to reduce friction for decision makers.
The page keeps strong calls to action, product qualification fields and secure form validation while preserving a clean enterprise layout.
It connects analog telephones, faxes and modems into SIP-based systems for cloud telephony, hosted UC, IP-Centrex, branch offices, SMBs and PSTN/TDM replacement projects.
Yes. The 4102S supports SIP over TLS, SRTP with AES 128-bit, TLS-encrypted provisioning and management, certificates, OCSP verification and TLS 1.3.
Voice codec support includes G.711 A-law/µ-law, G.726 and G.729a/b, with G.168 echo cancellation, DTMF handling, de-jitter buffer and packet loss concealment.
Yes. It supports real-time T.38 fax and clear-channel G.711 fax/modem pass-through for legacy analog data services over IP networks.
Management options include Web GUI, SSH, Telnet, SNMP v1/v2c/v3, TR-069/TR-104/TR-111, auto-provisioning through FTP/TFTP/HTTP/HTTPS and event notifications.
Networking includes IPv4/IPv6, multiple IP addresses per link or VLAN, multiple VLANs, DHCP client/server, PPPoE, IEEE 802.1x, LLDP-MED, QoS, NAT, firewall and static routing.
Use this form to route technical inquiries about 4102S deployment, SIP registration, FXS integration, fax/modem support, secure provisioning, NAT/firewall design or cloud telephony migration.