Grandstream Networks GHP6 Series Administration Manual page 15

Silence Suppression
Jitter Buffer Type
Jitter Buffer Length
Voice Frames Per TX
G.723 Rate
SRTP Mode
SRTP Key Length
Crypto Life Time
RTCP Mode
RTCP Keep-Alive Method
RTP Keep-Alive Method
If set to "Yes", when silence is detected, a small quantity of VAD packets (instead of audio
packets) will be sent during the period of no talking. For codec G.723 and G.729 only.
Default setting is "No".
Selects either Fixed or Adaptive for jitter buffer type, based on network conditions. The
default setting is "Adaptive".
Selects jitter buffer length from 100ms to 800ms, based on network conditions. The default
setting is "300ms".
Configures the number of voice frames transmitted per packet. It is recommended that the
IS limit value of Ethernet packet is 1500 bytes or 120 kbps. When configuring this, it should
be noted that the "ptime" value for the SDP will change with different configurations here.
This value is related to the codec used in the codec table or negotiate the payload type
during the actual call. For example, if set to 2 and the first code is G.729, G.711 or G.726,
the "ptime" value in the SDP datagram of the INVITE request is 20 ms. If the "Voice
Frame/TX" setting exceeds the maximum allowed value, the phone will use and save the
maximum allowed value for the selected first codec. It is recommended to use the default
setting provided, and incorrect setting may affect voice quality.
The default setting is 2.
Selects encoding rate for G723 codec.
RTP Settings
Enable SRTP mode based on your selection from the drop-down menu.
● No
● Enabled but Not forced
● Enabled and Forced
● Optional
The default setting is "No".
Allows users to specify the length of the SRTP calls. Available options are:
● AES 128&256 bit
● AES 128 bit
● AES 256 bit
Default setting is AES 128&256 bit
Enable or disable the crypto lifetime when using SRTP. If users set to disable this option,
phone does not add the crypto lifetime to SRTP header. The default setting is "Yes".
Configure RTCP port negotiation rules.
● Default: Use the traditional RTCP port, which is "RTP port+1".
● Negotiate RTCP Port: Use attribute RTCP to negotiate.
● RTCP Mux: The caller actively negotiates the RTCP port and indicates RTCP Mux at the
same time.
● RTCP Mux Only: The caller forces RTCP Mux, generated by the local media port only
apply for RTP port.
Configures the RTCP channel keep-alive packet type.
● Receiver Report: The RTCP channel will sends "receiver report+source
description+RTCP extension" as keep-alive data.
● Sender Report: The RTCP channel will sends "Sender report+source description+ RTCP
extension" as keep-alive data.
Configures the RTP channel keep-alive packet type.