User guide

Chapter 1: Overview
MAXCS ACC 6.7.1 Administration Manual 13
Bandwidth Control for VoIP Sessions – Each server can configure the maximum
concurrent VoIP sessions based on its Internet or intranet bandwidth. This feature is to
ensure that voice quality will not be impacted if too many VoIP sessions are connected
at the same time.
Codec Profile – Multiple codec profiles with different settings can be created and ap-
plied to different locations. Each profile can have a different codec, jitter buffer, and
packet length to accommodate different IP connections.
DNIS Name Display and Routing over IP Tie Trunk – allows for DNIS information
to be transferred to the other system when routed over IP tie-trunks. DNIS name of
matched entry can be displayed at AltiConsole, MaxCommunicator, MaxAgent, and
handset.
Caller ID/Name Sent Over IP Tie Trunk – SIP supports sending the caller’s name,
so SIP and H.323 calls may display different caller ID information.
DTMF payload embedded with RTP (RFC 2833) – this feature helps to resolve DTMF
tone detection and regeneration when using G.723.1 or G.729 codecs. Low bit rate
compression can distort DTMF tones during compression and cause the far end device to
not be able to recognize the DTMF digits. RFC 2833 specifies a separate RTP payload
format to carry DTMF information to ensure the other side can recognize the tone
properly.
Dynamic Jitter Buffer – due to various delays in the IP network, audio packet streams
may be delivered late or out of order. The system is able to buffer incoming packets and
re-sequence them by maintaining a queue. This queue is adjusted dynamically to accom
-
modate different network environment characteristics.
Echo Cancellation – due to bandwidth limitations and device loading, long delays may
occur during packet delivery process, which worsens the echo effect voice speech. Echo
cancellation is provided to maintain reasonable voice quality.
G.711 Codec – toll quality (64K) digital voice encoding, which guarantees
interoperability and better voice quality.
G.723.1 Codec – a dual rate audio encoding standard, which provides near toll quality
performance under clean channel conditions.
G.729 A+B Codec – speech data encoding/decoding standard of 8 Kbps.
Global IP Dialing Table – The IP Dialing Table is configured in Enterprise Manager. The
IP Dialing Table configuration is used to create location-based routing in the Enterprise.
H.323 Tie-Trunk Support – Ensures backward compatibility to systems using AltiGen’s
AltiWare versions prior to 5.1.
IP Extension Auto Failover – when an IP extension is unreachable, the system will
automatically fail over to a pre-configured Mobile Extension.
IP Group Paging – allows the use of voice paging to IP phone users in a group.
NAT Configuration for SIP/H.323 – When AltiServ is behind NAT with a private IP
address, this feature helps to resolve IP address resolution problems when communicat
-
ing with an external VoIP device.
Silence Detection and Suppression – when silence suppression is enabled and silence
is detected, the system stops sending packets to the other side. The other side does not
receive any packets and plays silence.
VoIP Hop-Off Call Support – allows an extension to access a PSTN trunk on the re-
mote system and “hop off” to dial an outside telephone number. This hop off feature can
be enabled or disabled on the remote system. Outcall restrictions for hop off calls are
configurable.