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Quick Overview

Mediatrix C730 4 FXO Ports - Multi-Function Gateway w/VoIP Analog Adapter, and QoS Control

 This product is a replacement for: Mediatrix 1204 4 Port FXO Gateway SIP 

Mediatrix C730 4 FXO Ports       

The Mediatrix C730 with 4 FXO Ports, are multi-function devices combining VoIP Analog Adapter, Gateway and QoS control in a secure and powerful platform. The C730 features 4 FXO interfaces, provides an ideal solution for enterprise voice applications or for connecting to a service provider’s broadband access. With FXO interfaces, it is the ideal solution to deploy private or hosted toll bypass networks. It provides a simple, transparent and cost-effective way of maintaining a connection to the PSTN. The Mediatrix C7 Series also allows Enterprises, Service Providers, and System Integrators to deploy secure systems and generate additional revenue streams.

The Mediatrix C730 enables cost-effective VoIP deployments in small offices for both IP Centrex and private network applications. It has the additional benefit of supporting high compression codecs simultaneously on each voice ports, thus saving valuable bandwidth. The Mediatrix C7 Series gateways make use of existing broadband access equipment to connect to any standards-based VoIP network. It also offers security features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.

The C7 Series is designed for:

• Enterprises with branch offices and remote offices, including SOHO.
• Enterprises seeking the benefits of Voice over IP without discarding existing legacy equipment.
• Enterprises seeking to implement Voice over IP through a gradual, transparent, and cost-effective migration.
• Multi-tenant / multi-dwelling units and premises seeking to implement Voice over IP while maintaining existing wiring and handsets.
• Enterprises looking to implement secure RTP (SRTP) and Secure SIP (SIP over TLS).

The Mediatrix C7 offers security features such as TLS, SRTP, certificate management, and HTTPS.



    IP Telephony Protocol
    SIP – RFC3261
    Multiple Virtual Gateways
    Multiple SIP Proxy support via DNS SRV
    Dual-Stack IPv4/IPv6 support using ANAT (RFC 4091)
    OCSP (Online Certificate Status Protocol) revocation status verificationfor TLS links
    ANAT grouping in the SDP
    Media stream can be on a different network than SIP signalling
    Call router automatic routes can have multiple SIP gateways for sources and destinations
    Voice Processing
    Vocoders: G.711 (A-law, μ-law), G.723.1, G.726, G.729a/b
    G.168 echo cancellation (64 ms)
    DTMF detection and generation
    Carrier tone detection and generation
    Silence detection / suppression and Comfort Noise Generation level software adjustable
    Configurable de-jitter buffer
    Configurable tones (dial, ringing, busy)
    Configurable transmit packet length
    RTP/RTCP - RFCs 1889 / 1890 / 2833 / 3389
    Enhanced Security
    HTTPS, for web pages and for exchange of Configuration File.
    SRTP with MIKEY and SDES: Supported Cypher, AES – 128 bits.
    MIKEY key management protocol (RFC 3830 and 4567).
    SDES key management protocol (RFC 4568).
    X.509 Certificate management.
    Fax and Modem Support
    Group 3/Super G3 Fax real-time FoIP over clear channel (G.711),
    G.726 or T.38
    T.38 Fax relay (9.6 k, 14.4 k)
    G.711 Fax and Modem Bypass
    T.38, fax tone detection and pass-through on G.711 and G.726
    Web-based GUI
    TFTP, HTTP configuration up- and download (Autoprovisioning)
    TFTP, HTTP firmware upgrade
    SNMPv1/v2/v3 agent (MIB II and private MIB)
    TR-069 for massive deployments (optional feature available at purchase time)
    Remote Services activation to purchase service licenses for deployed units, permitting to enable TR-069 feature in the field
    TR-104 to permit large provisioning systems to communicate with
    CPE on private networks
    Subscribes services for IMS: Transfer methods and real time media provisioning
    Call Routing
    Local switching
    Interface hunt groups
    Routing Criteria; Interface, Calling/called party number, Time of day, day of week, date
    Number manipulation functions; Replace numbers, Add/remove, digits, Multiple remote gateways, PLAR
    Call properties manipulations
    SIP header manipulations
    SIP Fallback on remote destinations - Redirect SIP calls to remote SIP Endpoints

    Using the Call Router with SIP Redirect 302 Moved Temp messages.
    Support of field Remote Party ID.
    Call Router enhancement to support Privacy Header Remote-Party-ID.
    Voice Signalling
    Session Description Protocol (SDP)
    Compliant with MMTEL requirements for 3GPP specification 24.615 for Call Waiting requests
    QoS Marking
    IEEE 802.1p/Q
    Network Connection
    2 x 10/100 BaseT Ethernet RJ-45 connectors
    Enhanced Telephony Features
    Call Forward / Call Transfer / Conference Call / Call Waiting support
    Inter-digit timer and IP dialing
    Echo Cancellation / Dynamic Jitter Buffer / Voice Activity Detection /
    Silence Suppression
    Message Waiting Indication, via FSK
    Flash hook event signaling
    Caller ID Generation (Name & Number) as per Bellcore DTMF or FSK
    Call Completion (CCNR / CCBS)
    Operating Environment
    Operating temperature: 00C to 400C
    Storage temperature: -200C to 700C
    Humidity: up to 85%, non-condensing
    Power Supply
    External 100-240 VAC power supply.
    4.4 cm x 28.3 cm x 16.2 cm (1.73 in. x 11.1 in. x 6.4 in)




MFG Part #: C730-01-MX-D2000- K-000


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