It has been extended for video conferencing, streaming media distribution, instant messaging, presence information, file transfer, Internet fax and online games. SIP was designed to provide a signaling and call setup protocol for IP-based communications supporting the call processing functions and features present in the public switched telephone network (PSTN) with a vision of supporting new multimedia applications. In June 2002 the specification was revised in RFC 3261 and various extensions and clarifications have been published since. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular networks. The protocol was standardized as RFC 2543 in 1999. SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996 to facilitate establishing multicast multimedia sessions on the Mbone. For the transmission of media streams (voice, video) the SDP payload carried in SIP messages typically employs the Real-time Transport Protocol (RTP) or the Secure Real-time Transport Protocol (SRTP). For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security (TLS). SIP is designed to be independent of the underlying transport layer protocol and can be used with the User Datagram Protocol (UDP), the Transmission Control Protocol (TCP), and the Stream Control Transmission Protocol (SCTP). Most commonly, media type and parameter negotiation and media setup are performed with the Session Description Protocol (SDP), which is carried as payload in SIP messages. SIP works in conjunction with several other protocols that specify and carry the session media. A call established with SIP may consist of multiple media streams, but no separate streams are required for applications, such as text messaging, that exchange data as payload in the SIP message. SIP is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE ( VoLTE). You can simply keep all your existing numbers.The Session Initiation Protocol ( SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP operators offer flexible, modular formulas tailored to your communication needs. You have a higher or lower number of employees with more or less communication needs? You can add or remove voice channels whenever you want. Our SBC infrastructure and fraud detection system ensure that all Dstny telephone services can run securely. The Quality of Service or QoS also ensures optimum management of your bandwidth: it gives priority to voice packets before data packets, as a result of which all your communication is of high quality.Ī SIP trunking system is scalable. Thanks to SIP’s flexibility, you only pay for what you use.Īnother feature of interest is that you can simply keep your existing infrastructure. Calls between different sites of the same company are actually free. In short, you’ll reduce your telecom budget because you benefit from the SIP operators’ attractive rates for domestic and international calls. Thanks to SIP trunking, you can channel your incoming and outgoing calls via an xDSL line, but without the costs of a classic subscription.

That way, you can use the broadband network for incoming and/or outgoing calls.

SIP trunking forms an IP connection between your company’s switchboard and an SIP operator.

Sip trunking