User's Manual 1336 Document #: LTRT-10632
Mediant 800B Gateway & E-SBC
of users, SIP authentication server for SBC users
Transport Mediation SIP over UDP/TCP/TLS/WebSocket, IPv4 / IPv6, RTP / SRTP (SDES/D
Message Manipulation Ability to add/modify/delete SIP headers and message body using advanced
regular expressions (regex)
URI and Number
Manipulations
URI user and host name manipulations, ingress and egress digit
manipulation
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, AMR-WB (G.722.2), SILK-NB/WB, Opus-NB/WB
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
NAT Local and far-end NAT traversal for support of remote workers
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
Maintains 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 with two-box redundancy, 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
Routing Features Least-cost routing, call forking, load balancing, E911 gateway support,
emergency call detection and prioritization
SIPRec IETF standard SIP recording interface
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