AudioCodes Mediant 800B User Manual page 998

Function
Interoperability
SIP B2BUA
SIP Interworking
Registration
Transport Mediation
Header Manipulation
URI and Number
Manipulations
Hybrid PSTN Mode
Transcoding and Vocoders
Signal Conversion
NAT
Voice Quality and SLA
Call Admission Control
Packet marking
Intelligent Voice
Stand Alone Survivability
Impairment Mitigation
Voice Enhancement
Voice Quality Monitoring
Quality of Experience
Test Agent
Media De-anchoring
SIP Routing
Routing Methods
User's Manual
Full SIP transparency, mature and broadly deployed SIP stack
3xx redirect, REFER, PRACK, Session Timer, Early media, Call hold,
Delayed offer
Registration and authentication on behalf of an IP-PBX
SIP over UDP to SIP over TCP or SIP over TLS, IPv4 to IPv6 ,v.34
fax, RTP to SRTP
Add/modify/delete SIP headers using advanced regular expressions
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:

Narrowband: SILK, G.711a/mu, G.723.1, G.729A/B, iLBC, AMR,
G.726

Wideband: G.722, AMR-WB, SILK WB
DTMF/RFC 2833, Inband/T.38 Fax, Packet-time Conversion, V.150.1
Local and Far-End NAT Traversal for support of remote workers
Based on bandwidth, session establishment rate, number of
connections/registrations
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).
Packet Loss Concealment, Dynamic Programmable Jitter Buffer,
Silence Suppression/Comfort Noise Generation, RTP redundancy,
broken connection detection
Transrating, RTCP XR, Acoustic echo cancellation
AudioCodes Session Experience Manager (SEM)
Access control and media quality enhancements based on QoE and
bandwidth utilization
Ability to remotely verify connectivity, voice quality and SIP message
flow between SIP UAs
Hair-pinning of local calls to avoid unnecessary media delays and
bandwidth consumption
Request URL, IP Address, FQDN, ENUM, advanced LDAP
998
Mediant 800B Gateway and E-SBC
Specification
Document #: LTRT-10286