S.I.P VoIP

The Voice over Internet Protocol (VoIP) is bringing about significant changes in the telecommunications industry with innovative services enabled by flexible and efficient packet transport.

At the heart of today’s VoIP networks is the Session Initiation Protocol, which is increasingly used for interworking functions between network services, as well as for signalling between devices. An increasing number of platforms now offer an extensive set of board and software components and voice coding technologies for developers designing S.I.P-based VoIP solutions.

More and more V.A.S platforms now require the capability for new S.I.P-based applications, such as audio and video conferencing, call forwarding or any number of enhanced call controls.

S.I.P provides the network control for IP sessions that is necessary for operators to build on the growing popularity of basic broadband telephony and VoIP services. S.I.P integration on V.A.S platforms provide the intelligence that frees operators to begin developing and launching sophisticated, personalised IP telephony services in the knowledge that they can be precisely controlled.

Subscribers who use S.I.P-based services are always in direct contact when they begin a call, or ‘session’, whether they use a S.I.P phone, laptop, mobile handset or PDA to initiate the session, and can also share multimedia content during the session. V.A.S operators using S.I.P open the way to operating cost benefits from the integration between basic IP telephony and fully interactive multimedia services while using the same IP infrastructure.

Telpla specialise in the following S.I.P specifications:

R.F.C 2246 Transport Layer Security (TLS) for S.I.P
R.F.C 2327 Session Description Protocol (SDP)
R.F.C 2976 S.I.P Info for digit transmission (#,*) and interworking DTMF
R.F.C 3261 S.I.P Basic
R.F.C 3262 S.I.P PRACK
R.F.C 3263 Locating S.I.P servers for DNS lookup SRV and A records
R.F.C 3264 SDP Offer/Answer Model
R.F.C 3265 S.I.P Subscribe/Notify
R.F.C 3311 S.I.P Update
R.F.C 3325 Asserted Identity
R.F.C 3326 S.I.P Reason Header
R.F.C 3372 S.I.P for Telephones (S.I.P-T)
R.F.C 3398 ISUP/S.I.P Mapping
R.F.C 3515 S.I.P Refer
R.F.C 3578 ISUP Overlap Signaling to S.I.P
R.F.C 3581 Symmetric Response Routing
R.F.C 3666 S.I.P to PSTN Call Flows
R.F.C 3725 Third Party Call Control for S.I.P
R.F.C 3764 ENUM for S.I.P Address of Record
R.F.C 4028 S.I.P Session Timer
R.F.C 4244 S.I.P History info (for call diversion)
R.F.C 4904 S.I.P tgrp (trunk group) parameter
S.I.P 3xx Gateway Responses and 302 Initiate
S.I.P Diversion Header
S.I.P Trunk Group IDs (OTG, DTG)
S.I.P Coder Negotiation
S.I.P Busy Out
ITU-T Q.1912.5 – S.I.P and ISUP Interworking (includes S.I.P-I) and Overlap signaling (S.I.P to S.I.P ISUP)
S.I.P mediation (S.I.P to S.I.P)
S.I.P to S.I.P-I/S.I.P-T