Grandstream Networks GXP2110 User Manual page 38

Refer-To Use Target
Contact
Transfer on Conference
Hangup
Preferred Vocoder
SRTP Mode
Symmetric RTP
Silence Suppression
Voice Frames per TX
No Key Entry Timeout
Grandstream Networks, Inc.
Default is NO. If set to YES,
uses the transferred target's Contact header information.
Defines whether or not the call is transferred to the other party if the initiator of the
conference hangs up.
Default setting is set to No.
GXP supports up to 7 different Vocoder types including G.711(a/µ) (also known
as PCMU/PCMA), G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band).
Configure Vocoders in a preference list that is included with the same preference
order in SDP message. Enter the first Vocoder in this list by choosing the
appropriate option in "Choice 1". Similarly, enter the last Vocoder in this list by
choosing the appropriate option in "Choice 8".
Enable SRTP mode based on selection. Default is No.
Selects whether or not symmetric RTP is supported.
This controls the silence suppression/VAD feature of the audio codec G.723 and
G.729. 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. If set to
"No", this feature is disabled.
This field contains the number of voice frames to be transmitted in a single
Ethernet packet (be advised the IS limit is based on the maximum size of
Ethernet packet is 1500 byte (or 120kbps)).
When setting this value, be aware of the requested packet time (ptime, used in
SDP message) is a result of configuring this parameter. This parameter is
associated with the first codec in the above codec Preference List or the actual
used payload type negotiated between the 2 conversation parties at run time.
E.g., if the first codec is configured as G.723 and the "Voice Frames per TX" is set
to 2, then the "ptime" value in the SDP message of an INVITE request will be
60ms because each G.723 voice frame contains 30ms of audio. Similarly, if this
field is set to 2 and the first codec is G.729 or G.711 or G.726, then the "ptime"
value in the SDP message of an INVITE request will be 20ms.
If the configured voice frames per TX exceeds the maximum allowed value, the IP
phone will use and save the maximum allowed value for the corresponding first
codec choice. The maximum value for PCM is 10 (x10ms) frames; for G.726, it is
20 (x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64
(x10ms) and 64 (x2.5ms) frames respectively.
Please be careful when editing these parameters. Adjusting these parameters will
also change the dynamic jitter buffer. The GXP has a patent dynamic jitter buffer
handling algorithm. The jitter buffer range is 20 ~ 200 ms.
Grandstream recommends using the default settings provided. Grandstream
does not recommend adjusting these parameters if you are an average user.
Incorrect settings will affect the voice quality. Please refer to the Codec FAQ at
http://www.grandstream.com/pdf/FAQ-Codec.pdf for more technical detail.
Default is 4 seconds.
GXP2110 User Manual
Firmware 1.0.0.44
then for Attended Transfer, the "Refer-To" header
Page 38 of 42
Last Updated: 09/2010