AltiGen comm ACM 5.1 - Manual (Page 344)

Chapter 25: Enterprise VoIP Network Management
330 AltiWare ACM 5.1 Administration Manual
Assigning Codec Profiles to IP Addresses
You can specify what codec profile to use when connecting to the following VoIP devices:
• IP phones on the LAN
• a remote IP phone over WAN
• a remote AltiGen system over WAN
• SIP Trunk service provider over WAN
• multiple AltiGateways on the LAN
The codec profile assigned in the IP Device Range table (shown below) supersedes the
codec profile defined in the IP dialing table if the IP address is duplicated in both tables.
G.711/G.723/G.729
Jitter Buffer Range
(ms) Indicates the delay, in milliseconds, used to buffer
G.711/G.723/G.729 voice packets received from the
IP network. Voice packets sent over the IP network
may incur different delays due to network load or
congestion. The jitter buffer helps to smooth out the
delay variation in the arriving voice packets and
maintain voice quality at the receiving end.
The default values for the jitter buffer for G.711 is 10
min. to 100 max milliseconds.
The default values for the jitter buffer for G.723 is 30
min. to 480 max milliseconds.
The default values for the jitter buffer for G.729 is 10
min. to 480 max milliseconds.
G.711 RTP Packet
Length (ms) Lets you configure the length of the RTP packets for
G.711 in milliseconds. The RTP packet length can be
set to 10, 20 or 30 milliseconds. The smaller the
packet length, the larger the bandwidth required.
G.729 RTP Packet
Length (ms)
Lets you configure the length of the RTP packets for
G.729 in milliseconds. The RTP packet length can be
set to 10, 20 or 30 milliseconds.
DTMF Delivery
(Applies to SIP protocol
only) Default—If SIP INFO is used to deliver DTMF.
RFC 2833—The DTMF pay load is embedded with
RTP. Most 3rd-party SIP gateways support this
standard. Applies to SIP TRUNK only.
In band—If deliver DTMF tone over the voice band.
It’s not reliable over G.711 codec and will not work
over G.729/G.723 codec
SIP Early Media
(Applies to SIP protocol
and SIP trunk only) SIP Early Media allows two SIP devices to
communicate before a SIP call is actually
established. It is important for interoperability with
the SIP trunk carrier’s PSTN gateway. If SIP Early
Media is not checked, the caller may not hear the
exact ringback tone provided by the CO (the caller
may not hear any ringback tone at all).
Parameter Description