User guide

Chapter 25: Enterprise VoIP Network Management
304 MAXCS ACC 6.7.1 Administration Manual
Name
Name of the codec profile. You can modify the name, and click
Apply. The Default profile name cannot be changed.
Codec
There are several options:
• G.711 Mu-Law
• Prefer G.723.1, support G.729
• Prefer G.729, support G.723.1
• G.711 A-Law
• Prefer G.711 Mu-Law, support G.711 A-Law
• Prefer G.711 A-Law, support G.711 Mu-Law
G.711 provides toll quality digital voice encoding, and G.723
and G.729 use low rate audio encoding to provide near toll
quality performance under clean channel conditions.
G.711/G.723/G.729
Silence Suppression
When silence suppression is enabled, and silence is detected
during a call, MAXCS stops sending packets to the other side.
This decreases the bandwidth requirement, however the voice
quality may be degraded slightly. These are system-wide
settings.
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.
In band – If DTMF tone is delivered over the voice band. It’s
not reliable over G.711 codec and will not work over G.729/
G.723 codec
Parameter Description