Aria Communications SIP Trunking
What is SIP?
The Session Initiation Protocol (SIP) is a communications protocol for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, as well as in instant messaging over Internet Protocol(IP) networks.
The protocol defines the methodology of SIP communications and the specific format of messages exchanged for cooperation of the participants in multimedia sessions. A call established with SIP may consist of multiple media streams, but none are required in text messaging, for which the payload is carried directly in the SIP message. 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). It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).
SIP works in conjunction with several other protocols that specify and carry the session media. Media type and parameter negotiation and media setup is performed with the Session Description Protocol (SDP), which is carried as payload in SIP messages. For the transmission of media streams (voice, video) SIP 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).
What are SIP Trunks?
SIP Trunks are virtual connections between your IP-PBX and our VoIP network. SIP or the Session Initiation Protocol, is the standard communications protocol for voice and video in a Unified Communications (UC) solution using an internet or broadband connection. A SIP trunk replaces the need for traditional analog service that utilises the PSTN or POTS (plain old telephone system).
A SIP trunk allows connections to scale up or down in real-time based on channels. Channels is simply another way of describing concurrent calls or the number of lines for voice or video. The only major consideration in understanding a channel limit is the bandwidth available on the internet connection at any one time.
Aria SIP Powered by VoiceHost SIP Trunks are RFC3261compliant.