NetBorder Transcoding Gateway

    Netborder Transcoding Gateway

     

    Product Overview 

    The NetBorder Transcoding Gateway is an IP to IP VoIP transcoding and media server.  It operates as an intelligent SIP back to back user agent to provide any to any audio codec transcoding and media processing while allowing business logic to remain in the Soft Switch.

    Here is a high level list of the functions that could be performed by this product to fix the various media incompatibilities that could be found by two VoIP endpoints.

    • Any to Any voice codec conversion (ex: G729, G722, G711...) 
    • RTP packet size conversion
    • RTP DTMF event (RFC4733) to/from inband DTMF
    • Noise Reduction *
    • Voice Quality Enhancement *


    netborder-trancoding-gateway-carrier-provider-1.jpg

    Who is the Netborder Transcoding Gateway For?

    • Carriers & Service Providers
    • SIP Trunk Providers
    • Hosted PBX Providers
    • Call Centres
    • Large Enterprise
       

    Product Architecture

    The following diagram shows the logical interfaces of the appliance.

    • GUI Interface
      Used to Configure, Manage and Operate Netborder Transcoding Gateway
    • Intelligent SIP Back to Back User Agent 
      Transcode any to any codec while allowing business logic to remain in the Soft Switch.
    • Proxy Like Features
      SIP Header and SDP fidelity with Soft Switch controlled SDP codec selection. 
    • First class technical support
      To help you integrate the Netborder Transcoding Gateway in your Network.
       

    netborder-trancoding-gateway-carrier-provider-prod-arch.jpg




    Product Scenarios

    •  Incoming Call from ISP/Telco to your network

      netborder-trancoding-gateway-carrier-provider-inbound-call.jpg netborder-trancoding-gateway-carrier-provider-inbound-call.jpg  netborder-trancoding-gateway-carrier-provider-inbound-call-detailed.jpg

    • Outgoing Call from your network to ISP/Telco
       
      netborder-trancoding-gateway-carrier-provider-outbound-call.jpg   netborder-trancoding-gateway-carrier-provider-outbound-call-detailed.jpg

     

     

     

    Product Features

    Audio Codecs

    • G711 PCMU and PCMA (10, 20, 30ms packet)
    • G723.1 (30ms, packet)
    • G726 16, 24, 32, 40 Kbps (20, 30ms packet)
    • G729AB (10, 20, 30ms packet)
    • iLBC 13.3 (30ms packet), 15.2Kbps (20ms packet)
    • AMR 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.20, 12.20 Kbps (20, 40ms packet)
    • G722 (20, 30ms packet)
    • G722.1 Siren 7 (20ms packet)

    Media Mediation

    • Audio Codec mediation
    • Packet Size mediation
    • In-band DTMF to/from RFC2833 events

    Advanced Voice Features

    • Comfort Noise Generation (CNG)
    • Voice Activity Detection (VAD)
    • Packet Loss Concealment (PLC)
    • Noise Reduction *
    • Voice Quality Enhancement *

    Media Physical Interfaces

    • Up to 2x1Gbe interfaces (when populated with 2 D500 media adapters)
    • IPv4 transport
    • Requires up to 5 Static IP addresses per D500 media adapters

    Control/Management Physical Interfaces

    • 2x1Gbe interfaces
    • IPv4 transport
    • 1 Static IP address per interface

    SIP Call Control

    • SIP B2BUA (internal/external SIP profiles)
    • Configurable per SIP profile
      • Transports UDP, TCP or TCP+UDP
      • SIP session timer
      • Inbound/Outbound Audio Codec list
      • Option SIP Proxy
        • Single user Registration and authentication
    • Support to specify outbound audio codec list per call via SIP-X header 
    • Supported RFCs
      • RFC3261 - SIP: Session Initiation Protocol
      • RFC3262 - Reliability of Provisional Responses in the Session Initiation Protocol (SIP) (PRACK)*
      • RFC3264 - An Offer/Answer Model with the Session Description Protocol (SDP)
      • RFC4028 - SIP Session Timer
      • SIP HTTP Authentication
      • SIP Re-INVITE (stop on the receiving call leg)

    Media Transport

    • RTP/RTCP as per RFC3550
    • DTMF relay as RFC2833
    • Adaptive Jitter Buffer

    Capacity

    • Minimum:  400 sessions
    • Maximum: 4000 session 
    • Note: 
      • Capacity is based on G711 to/from G729 transcoding sessions (number of transcoding sessions will be lower other combination of codecs)  
      • Each session contains 4 legs to complete a full duplex call.

    Configuration and Management

    • WEB UI (port 81)
      • Authentication access
      • Control/Management Interfaces configuration and monitoring
      • Media Interfaces configuration and monitoring
      • SIP profiles configuration
      • Audio profiles configuration
      • Transcoding resources statistics
      • System resources monitoring (CPU/memory/network activity)
      • View logs
      • Backup/Restore Configuration
      • Supported WEB browsers (Internet Explored 7.0-9.0, Firefox 10.0, Chrome, Safari)
    • SSH access for Sangoma support only
    • Detailed logs with configurable file size with rollover supports

    Support and Professional Services 

    • Sangoma Engineers are here to support your success. 
    • Whether you need technical support and software maintenance, training, consultation and installation services, Sangoma can help you. 
    • Please contact your Sales representative for more information.