User's Manual 1068 Document #: LTRT-89730
Mediant 3000
Function Specification
Transcoding and Vocoders
Coder normalization including transcoding, coder enforcement and
re-prioritization, extensive vocoder support: G.711, G.723.1, G.726,
G.729, GSM-FR, AMR-NB/WB, SILK-NB/WB, Opus-NB/WB, iLBC,
QCELP, GSM EFR, EVRC, MS-RTA NB/WB, SPEEX NB/WB
NAT
Local and far-end NAT traversal for support of remote workers
Signal Conversion
DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, V.34, packet-time
conversion, V.150.1
WebRTC Controller
Interworking between WebRTC devices and SIP networks
Supports WebSocket, Opus, VP8 video coder, lite ICE, DTLS, RTP
multiplexing, secure RTCP with feedback
Voice Quality and SLA
Call Admission Control
Based on bandwidth, session establishment rate, number of
connections/registrations
Packet Marking
802.1p/Q VLAN tagging, DiffServ, TOS
Standalone Survivability
Maintain local calls in the event of WAN failure, outbound calls can
use PSTN fallback for external connectivity (including E911).
Impairment Mitigation
Packet Loss Concealment, Dynamic Programmable Jitter Buffer,
Silence Suppression/Comfort Noise Generation, RTP redundancy,
broken connection detection
Voice Enhancement
Transrating, RTCP-XR, acoustic echo cancellation, replacing voice
profile due to impairment detection, fixed & dynamic voice gain
control
Direct Media (No Media
Anchoring)
Hair-pinning of local calls to avoid unnecessary media delays and
bandwidth consumption
Voice Quality Monitoring
RTCP-XR, AudioCodes Session Experience Manager (SEM)
High Availability
(Redundancy)
SBC high availability, active calls preserved
Quality of Experience
Access control and media quality enhancements based on QoE and
bandwidth utilization
Test Agent
Ability to remotely verify connectivity, voice quality and SIP message
flow between SIP UAs
SIP Routing
Routing Methods
Request URL, IP Address, FQDN, ENUM, advanced LDAP, third-
party routing control through REST API
Advanced Routing Criteria
QoE, bandwidth, SIP message (SIP request, coder type, etc.),
Layer-3 parameters
Redundancy
Detection of proxy failures and subsequent routing to alternative
proxies
Routing Features
Least-cost routing, call forking, load balancing, E911 gateway
support, emergency call detection and prioritization
SIPRec
IETF standard SIP recording interface
Management
OAM&P
Browser-based GUI, CLI, SNMP, EMS, INI Configuration file, REST
To Next Page
Loading...
Need help?
General
