BeoX is one of the largest VoIP carriers of international voice traffic in the world and a leader in direct voice termination. Founded in 2004, our company has presence in the USA, UK and Turkey and supports infrastructure around the globe.
BeoX have been a major player in the international voice termination segment for the last six years and have created an extensive portfolio of international voice termination products designed specifically to suit our partners’ different needs.
Our core competence lies in our ability to offer the high quality voice termination at best possible price brackets by leveraging our optimum use to the latest technologies, our direct termination experience, our vast network of in-country termination partners and, the economies of scale our significant volumes of traffic yield.
Sales Enquiry : sales@beox.com
SIP is independent from the underlying transport protocol. It runs on the Transmission Control Protocol (TCP), the User Datagram Protocol (UDP) or the Stream Control Transmission Protocol (SCTP).[9] SIP can be used for two-party (unicast) or multiparty (multicast) sessions.
SIP employs design elements similar to the HTTP request/response transaction model.[10] Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format.
Each resource of a SIP network, such as a user agent or a voicemail box, is identified by a uniform resource identifier (URI), based on the general standard syntax[11] also used in Web services and e-mail. The URI scheme used for SIP is
sip: and a typical SIP URI is of the form: sip:username:password@host:port. If secure transmission is required, the scheme sips: is used and mandates that each hop over which the request is forwarded up to the target domain must be secured with Transport Layer Security(TLS). The last hop from the proxy of the target domain to the user agent has to be secured according to local policies. TLS protects against attackers who try to listen on the signaling link but it does not provide real end-to-end security to prevent espionage and law enforcement interception, as the encryption is only hop-by-hop and every single intermediate proxy has to be trusted.
SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP clients typically use TCP or UDP on port numbers5060 and/or 5061 to connect to SIP servers and other SIP endpoints. Port 5060 is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used for traffic encrypted with Transport Layer Security (TLS). SIP is primarily used in setting up and tearing down voice or video calls. It also allows modification of existing calls. The modification can involve changing addresses or ports, inviting more participants, and adding or deleting media streams. SIP has also found applications in messaging applications, such as instant messaging, and event subscription and notification. A suite of SIP-related Internet Engineering Task Force (IETF) rules define behavior for such applications. The voice and video stream communications in SIP applications are carried over another application protocol, the Real-time Transport Protocol (RTP). Parameters (port numbers, protocols, codecs) for these media streams are defined and negotiated using the Session Description Protocol (SDP), which is transported in the SIP packet body.
A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN). SIP by itself does not define these features; rather, its focus is call-setup and signaling. The features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ringback tones or a busy signal - are performed by proxy servers and user agents. Implementation and terminology are different in the SIP world but to the end-user, the behavior is similar.
SIP-enabled telephony networks can also implement many of the more advanced call processing features present in Signaling System 7 (SS7), though the two protocols themselves are very different. SS7 is a centralized protocol, characterized by a complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a client-server protocol, however most SIP-enabled devices may perform both the client and the server role. In general, session initiator is a client, and the call recipient performs the server function. SIP features are implemented in the communicating endpoints, contrary to traditional SS7 features, which are implemented in the network.
SIP is distinguished by its proponents for having roots in the IP community rather than in the telecommunications industry. SIP has been standardized and governed primarily by the IETF, while other protocols, such as H.323, have traditionally been associated with the International Telecommunication Union (ITU).
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