Design a SIP Switching system.

    To develop telephony applications such as SIP dialer, Operator panel, IVR, CTI, etc. on IP based open source switching system to meet common requirements of unified communication system.

    To develop VOIP management and performance reporting solution that helps to monitor performance and statistics

    such as call history ,call details ,QoS parametrs (jitter, packet loss, latency etc).

    Duplication of Database and realtime switching States to create" hot stand".

    Computational model for Hardware calibration for different VoIP applications based on BHCA and BHCC

    Testing and Certification for SIP RFCs.

    Implement call Admission control and alternate routing.

    Implement advanced routing techniques and realtime path replacement to ensure QOS.

Knowledge of Data Networking

IP V6 implementation.

Hot standby server for high availability.

BHCA/BHCC calculations & assessment of hardware requirement based on these calculations.

Call Admission Control-implementation.

QOS,Jitter packet, loss & bandwidth choking parameters to be factored.

Path replacement while call is in progress in case of deterioration.

Knowledge ITU Standards of Telephony

H323 implementation.

Q-sig implementation. Understanding Q-sig which is an ISDN protocol and integrate it with Freeswitch.

Video Conferencing is to be implemented on sip as well as H323 video phones.

SIP Standards & Compliance

Understanding each RFC.
Understanding resources for DTMF dialler & receiver/RBT, engage tone, Dial Tone that are system generated. Resource for codec allocation.Implement available codecs like 729/723 & any proprietary codec.
Key telephone with special features/use of extended SIP protocol and its implementation on Freeswitch.
Select SIP client from open source and implantation of various features on the same including making it work like an operator console with DSS / BLF / AB / AT / AS / docket less booking.

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