Brekeke Software Interoperable with Polycom KIRK Wireless Server 300 Brekeke Software, a developer of SIP telephony software for enterprises and service providers, has announced the confirmation of the interoperability between Polycom KIRK Wireless Server 300 (KWS300) with… →
Polycom Takes Wraps Off New Channel Program Polycom has rolled out a new partner program that the company hopes will elevate partner commitment and drive partners to attain specializations in skills in areas such as unified communications (UC) and… →
Polycom Revamps Partner Program To Challenge Cisco-Tandberg Polycom has revamped its solution provider partner program in an attempt to drive more growth with its most invested partners and capitalize on channel discontent stemming from Cisco (NSDQ:CSCO)'s… →
SIP (Session Initiation Protocol)Sip - is a signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol. Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information and online games. The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams, etc.
SIP was developed by the Internet Engineering Task Force (IETF) and was selected by 3GPP as a standard for IP Multimedia (IMS) in Release 5.
IAX (Inter-Asterisk eXchange protocol) Inter-Asterisk eXchange Protocol, or IAX, is the open-source protocol, developed for Asterisk, that's used to make connections between Asterisk servers. It also enables connections from Asterisk servers to people that are using the IAX protocol at home or in their office. Usually though, the term "IAX" is used in reference to IAX2, which is the 2nd, and most recent, version of the IAX protocol.
Although the IAX 2 signaling protocol is not an industry standard, like SIP, it performs basically the same functions and is rapidly becoming more robust due to the open-source nature of its coding. Additionally, IAX 2 is much easier to implement in situations where NAT (Network Address Translation) is required.
MGCP (Media Gateway Control Protocol) MGCP is a protocol used for controlling Voice over IP (VoIP) Gateways from external call control elements. MGCP is the emerging protocol that is receiving wide interest from both the voice and data industries. MGCP is defined in an informational (non-standard) IETF document, RFC 3435 , which obsoletes an earlier definition in RFC 2705 . It superseded the Simple Gateway Control Protocol (SGCP) .
MGCP is a protocol for controlling media gateways from call agents. In a VoIP system, MGCP can be used with SIP or H.323. SIP or H.323 will provide the call control functionality and MGCP can be used to manage media establishment in media gateways.
H.323 H.323 is a protocol standard for multimedia communications. H.323 was designed to support real-time transfer of audio and video data over packet networks like IP. The standard involves several different protocols covering specific aspects of Internet telephony. The International Telecommunication Union (ITU-T) maintains H.323 and these related standards.
Most voice over IP (VoIP) applications utilize H.323. H.323 supports call setup, teardown and forwarding/transfer. Architectural elements of a H.323 based system are Terminals, Multipoint Control Units (MCUs), Gateways, an optional Gatekeeper and Border Elements. Different functions of H.323 run over either TCP or UDP. Overall, H.323 competes with the newer Session Initialization Protocol (SIP), another proven standard often found in VoIP systems.
Voip Providers by Protocol
