AudioCodes Mediant 800B User Manual page 1336

Authentication
Transport Mediation
Message Manipulation Ability to add/modify/delete SIP headers and message body using advanced
URI and Number
Manipulations
Transcoding and
Vocoders
Signal Conversion
WebRTC Controller
NAT
Voice Quality and SLA
Call Admission Control Based on bandwidth, session establishment rate, number of
Packet marking
Standalone
Survivability
Impairment Mitigation
Voice Enhancement
Direct Media
(No Media Anchoring)
Voice Quality
Monitoring
High Availability
(Redundancy)
Quality of Experience Access control and media quality enhancements based on QoE and
Test agent
SIP Routing
Routing Methods
Advanced Routing
Criteria
Routing Features
SIPRec
User's Manual
of users, SIP authentication server for SBC users
SIP over UDP/TCP/TLS/WebSocket, IPv4 / IPv6, RTP / SRTP (SDES/DTLS)
regular expressions (regex)
URI user and host name manipulations, ingress and egress digit
manipulation
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, AMR-WB (G.722.2), SILK-NB/WB, Opus-NB/WB
DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, V.34, packet-time conversion,
V.150.1
Interworking between WebRTC devices and SIP networks Supports
WebSocket, Opus, VP8 video coder, lite ICE, DTLS, RTP multiplexing,
secure RTCP with feedback
Local and far-end NAT traversal for support of remote workers
connections/registrations
802.1p/Q VLAN tagging, DiffServ, TOS
Maintains local calls in the event of WAN failure. Outbound calls can use
PSTN fallback for external connectivity (including E911)
Packet Loss Concealment, Dynamic Programmable Jitter Buffer, Silence
Suppression/Comfort, Noise Generation, RTP redundancy, broken
connection detection
Transrating, RTCP-XR, Acoustic echo cancellation, replacing voice profile
due to impairment
detection, Fixed & dynamic voice gain control
Hair-pinning of local calls to avoid unnecessary media delays and bandwidth
consumption
RTCP-XR, AudioCodes Session Experience Manager (SEM)
SBC high availability with two-box redundancy, active calls preserved
bandwidth utilization
Ability to remotely verify connectivity, voice quality and SIP message flow
between SIP UAs
Request URL, IP address, FQDN, ENUM, advanced LDAP, third-party
routing control through REST API
QoE, bandwidth, SIP message (SIP request, coder type, etc.), Layer-3
parameters
Least-cost routing, call forking, load balancing, E911 gateway support,
emergency call detection and prioritization
IETF standard SIP recording interface
Mediant 800B Gateway & E-SBC
1336 Document #: LTRT-10632