Audiocodes Mediant 600

Audiocodes Mediant 600


VoIP Media Gateway Mediant 600 is a cost-effective product from AudioCodes, based on the advanced technology of voice transmission over packet networks, allows deploying reliable, cost-effective next-generation networks. Equipped with a compact 1U body, it is specially designed for interoperability of TDM and IP networks in enterprises and small access points.

Mediant 600 is based on VoIPerfect ™ technology, which is the underlying technology of all AudioCodes products and provides support for Multiple Vocoders, Long-tail echo cancellation, Adaptive Jitter Buffer, Packet Loss Reconstruction, etc.
The media gateway Mediant 600 uses processing technologies and transmission of the highest quality of voice for connecting traditional phones and PBX to IP-based networks and connects IP-PBX to the PSTN. It uses the security mechanisms and secures connections at all levels:
SRTP (media traffic) HTTPS, TLS (signalling) IP-SecAES, Access Lists, SNMPV3 (management)
Configuration, management, and control of the Mediant 600 can be personalized with the SNMP console or the data network through its own interface or dedicated centralized management of the entire product family of AudioCodes EMS.
Compact Modular Media Gateway Mediant 600 supports 1 or 2 E1/T1/J1 (CAS, EDSS, QSIG) interfaces or 4 to 8 digital basic rate BRI (2B + D) Euro-ISDN interfaces in accordance with ITU-T I .430 on the network (NT) or user (TE) sides.
Mediant 600 is fully compatible with the absolute majority of gateways, softswitches, gatekeepers, application servers and media servers, IP phones, session border controllers, firewalls and other manufacturers.

Providers in database: 6433
Register VoIP Provider

RTP Delivers Multimedia Data In Real Time

RTP Delivers Multimedia Data In Real TimeA Real-time Transport Protocol (RTP) is a protocol standard defined by the Internet Engineering Task Force (IETF) that outlines a management system for programs using real-time transmission of data through mutlicast or unicast network services in multimedia situations. Originally designed by the IETF for videoconferencing for multiple participants, the protocol is largely used in Voice over Internet Protocol applications. Despite its name, the protocol cannot guarantee real-time delivery of data, however, the RTP does compensate for jitters and can detect when data arrives out of sequence, both issues being popular in VoIP communication. In IP telephony, RTP works with a signaling protocol, such as SIP or H.323, in order to set up connections in a network.

View more WIKI
United States TinCanTalk

VOIP Service Provider - Business & Residential
Hosted PBX

Read more