HD voice Mobile and PC Softphone

PC and mobile Dialer subsystem allows VoIP Service Providers offer the best voice calling feature to their customers.

With increasing penetration of Internet VOIP has emerged as a cost effective alternative to the Voice communication. VOIP can deliver good voice quality if Internet Bandwidth is ample and network is reliable but in the places where Internet BW is less than required and there is packet loss and delay in the network, traditional VOIP cannot assure customer satisfaction as these conditions can make real time two way communication impossible .

Why use Bridge Dialer?

  1. Offers excellent HD audio calling feature
  2. Shows account balance information on mobile screen and also plays via IVR
  3. Supports wide range of Device models and Mobile operating system.
  4. Equipped with Adaptive jitter buffer, Noise reduction and Echo cancellation.
  5. Pass calls through firewalls through corresponding server counterpart.
  6. Can be private labeled

Components

Encryption and RTP Server

  • Supports clusters and new hardware can be added without disturbing the live
    traffic.
  • Modular approach makes it highly scalable.
  • In case of media IP change, dialers do not need to be reconfigured.
  • In built Fail-over server support
  • In built Load balance support
  • Flexible far-end NAT traversal.
  • Grace-full restart feature makes sure that in case of unavoidable restarts during traffic time do not create inconsistency in billing. In most of the cases even connected calls will be connected.
  • 8- Minimum hardware requirement :Quad Core Xeon 3.0 GHz,4 GB RAM, 80 GB SAS HDD (For approx 1000 calls)
  • 9- Software requirement : Cent OS (Linux) 64 bit

Soft Phone

  • Real-Time Balance Display
  • Balance deduction after the call
  • Call Timer clock display
  • Last 20 Calls CDR history in real-time itself
  • Microphone & Speaker Volume control
  • PC to Phone, Phone to PC & PC to PC Calling
  • Authentication – HTTP Basic, Digest Authentication
  • SIP UDP Support
  • Codec negotiation as per defined sequence by Service provider.
  • Skins supported
  • Phone book
  • Make installation packet
  • Works with all kind of Internet connections
  • Adaptive jitter buffer
  • Packet loss concealment for voice
  • Automatic Gain Control (AGC) for voice
  • Voice Activity Detection (VAD)
  • Acoustic Echo Cancellation (AEC) and support for speakerphone
  • Codecs Supported: OPUS, ISAC, G.729, iLBC, G.711ulaw and G.711alaw
  • Works with Wifi and 3G
  • Supported OS are MS Windows(2000/XP/2003/Vista/Win7)
  • Supported SIP Methods – REGISTER ,INVITE,CANCEL, INFO,BYE, ACK, REFER, SUBSCRIBE,OPTIONS, NOTIFY,MESSAGE, UPDATE
  • RFC supported – RFC 3261, RFC 3665, RFC 2833, RFC 2327, RFC 3264, RFC 3550, RFC 3263, RFC 3891,RFC 3515, RFC 3420, RFC 3892, RFC 3265, RFC 3666, RFC 3489, RFC 3920, RFC 3921, RFC 3922, RFC 3923, RFC 4622, RFC 4854, RFC 4979, RFC 3842,
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