AudioCodes Mediant 800B User Manual page 884

Function
Interoperability
SIP
ITSP and PBX support
Transport Mediation
Header Manipulation
URI and Number
Manipulations
Hybrid PSTN Mode
Transcoding and Vocoders
Signal Conversion
NAT
Signal Detection
Voice Quality and SLA
Call Admission Control
Packet marking
Intelligent Voice
Stand Alone Survivability
Impairment Mitigation
Voice Enhancement
Gain Control
Media Anchoring
SIP Routing
Routing Methods
User's Manual
Standalone SIP B2BUA, Netann (RFC4240), MSCML (RFC5022) or
RFC 4117 transcoding device control. Full SIP transparency, mature
& broadly deployed SIP stack
Interoperable with many SIP trunk Service Providers and PBX
vendors
SIP over UDP to SIP over TCP or SIP over TLS, IPv4 to IPv6 ,v.34
fax, RTP to SRTP
Programmable header manipulation. Ability to add/modify/delete
headers
URI User and Host name manipulations. Ingress & Egress Digit
Manipulation
Connect to TDM PBXs or PRI/CAS trunks for least-cost routing or
fallback. Also useful for gradual enterprise migration to SIP, Support
for analog and T1/E1/J1
Coder normalization, including: transcoding, coder enforcement and
re-prioritization. Extensive vocoder support: Wireline:
SILK, G.711a/mu, G.723.1, G.729A/B/E, Wideband: G.722, AMR-
WB, SILK WB and AMR, G.726
DTMF/RFC 2833, Inband/T.38 Fax, Packet-time Conversion,
V.150.1
Local and Far-End NAT Traversal for support of remote workers
DTMF/RFC2833, Packet-time conversion
Deny excessive calls based on session establishment rate, number
of connections and number of registrations (per SIP trunk or routing
domain)
802.1p/Q VLAN tagging, DiffServ, TOS
Multiple queues for granular prioritization of VoIP over other non-real
time traffic types, Integrated Queuing and scheduling schemes (Strict
Priority, Class based Prioritization queuing, fairness)
Maintain local calls in the event of WAN failure. Outbound calls use
PSTN Fallback for external connectivity (including E911). SAS
ensures call continuity between LAN SIP clients upon connectivity
failure with IP Centrex services (e.g., WAN IP PBX).
Packet Loss Concealment, Dynamic Programmable Jitter Buffer,
Silence Suppression/Comfort Noise Generation, RTP redundancy,
broken connection detection
Transrating, RTCP XR
Fixed & dynamic voice gain control
Hair-pinning of local calls to avoid unnecessary media delays and
bandwidth consumption
Request URL, Source/ Destination IP Address, Fully Qualified
884
Mediant 800B GW & E-SBC
Specification
Document #: LTRT-10274