AltiGen comm ACM 5.1 manual Assigning Codec Profiles to IP Addresses, RTP Packet, Length ms

Chapter 25: Enterprise VoIP Network Management

Parameter	Description
G.711/G.723/G.729	Indicates the delay, in milliseconds, used to buffer
Jitter Buffer Range	G.711/G.723/G.729 voice packets received from the
(ms)	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	Lets you configure the length of the RTP packets for
Length (ms)	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	Lets you configure the length of the RTP packets for
Length (ms)	G.729 in milliseconds. The RTP packet length can be
	set to 10, 20 or 30 milliseconds.
DTMF Delivery	Default—If SIP INFO is used to deliver DTMF.
(Applies to SIP protocol	RFC 2833—The DTMF pay load is embedded with
only)	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	SIP Early Media allows two SIP devices to
(Applies to SIP protocol	communicate before a SIP call is actually
	established. It is important for interoperability with
and SIP trunk only)
	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).

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.

330AltiWare ACM 5.1 Administration Manual