Capacity and Voice Processing
51
SmartNode 10100 Series User Manual
B • Specifications
Capacity and Voice Processing
VoIP channels
—128 to 256
PSTN interfaces
—Dual RJ48C for BITS or T1/E1 for signaling
VoIP interfaces
—Dual 100/1000Base-T • RJ45 connectors on rear of unit
Vocoding
—Universal codecs: G.711, G.723.1, G.726, G.729ab, T.38 • Other codecs: G.722.2 (AMR-WB),
G.728, G.729eg, iLBC, clear mode (RFC 4040)
Fax/Modem/Data
—T.38 fax relay (V.17 and V.34) • Automatic G.711 fallback • Modem and data pass-
through
DTMF relay
—RFC 2833, SIP INFO Method, In-band
Echo cancellation
—G.168 echo cancellation • 128 ms echo tail on all channels simultaneously
Voice processing
—Dynamic and programmable jitter buffer (20 to 200 ms) • Voice activity detection (VAD) •
Comfort noise generation (CNG)
Management interfaces
—Dual 100/1000Base-T for OAM&P
Simultaneous Signaling Support
SIP
—Supported RFCs: 2327, 2976, 3261, 3262, 3263, 3264, 3311*, 3323*, 3325*, 3398, 3515*, 3578*,
3764, 3891, 4028 (*partial compliance)
SIGTRAN
—M2PA, M1UA, M3UA, IUA • SS7 termination and/or relay supported
SS7
—Up to 64 x MTP2 links (56, 64, n x 56/64 kbps, HSL) • Multiple redundant MTP2 links • Up to 64
MTP3 originating point codes and linksets • ISUP variants: ITU 92, ITU 97, ANSI 88, ANSI 92, ANSI 95,
Telcordia 97, ETSIv2, ETSIv3, China, Singapore, UK Brazil
ISDN PRI
—Q.931 ISDN PRI: NI-2, 4ESS, 5ESS, DMS-100, DMS-250, Euro ISDN ETSI NET5 (France,
Germany, UK, China, Hong Kong, Korea), NTT (Japan), Australia
CAS
—MFC R2 (standard ITU, Brazil) • Customizable protocol script files
SmartNode-CONTROL
Standalone call control
—Any to any call routing (TDM-VoIP, TDM-TDM, VoIP-VoIP with transcoding) •
Call routing based on: trunk group, calling/called numbers, nature of address, ASR, time of day, load-based,
cost-based, TO:, FROM: Request URI, redirect numbers, and other parameters • NPA-NXX routing (100k+
table entries, Excel or CVS file upload) • Route retries • Call transfer (REFER, AT&T TR 50075)
H.248 (MEGACO) call control
—ITU-T H.248 versions 1 and 2 • UDP, SCTP, IPsec transport • DTMF and
fax detection • DTMF, announcements and call progress tone generation • Call quality and inactivity alerts
Session management and billing
—SIP peer availability polling • RTP inactivity monitoring • CDR generation
(RADIUS and text file) OAM&P
