Account/Codec Settings - Grandstream Networks WP820 Administration Manual

Sets which party will refresh the active session if the phone makes outbound calls. If it is set to "UAC" and the remote
UAC Specify party does not support Refresher feature, the phone will refresh the active session. If it is set to "UAS", the remote party
Refresher will refresh it. If it is set to "Omit", the header will be omitted so that it can be selected by the negotiation mechanism. The
default setting is "Omit".
Specifies which party will refresh the active session if the phone receives inbound calls. If it is set to "UAC", the remote
UAS Specify party will refresh the active session. If it is set to "UAS" and the remote party does not support refresh feature, the phone
Refresher will refresh it.
The default setting is "UAC".
Caller Sets the caller party to act as refresher by force. If set to "Yes" and both party support session timers, the phone will enable
Request the session timer feature when it makes outbound calls. The SIP INVITE will include the content "refresher=uac". The
Timer default setting is "No".
Callee Sets the callee party to act as refresher by force. If set to "Yes" and both parties support session timers, the phone will
Request enable the session timer feature when it receives inbound calls. The SIP 200 OK will include the content "refresher=uas".
Timer The default setting is "No".
Configures the session timer feature on the phone by force. If it is set to "Yes", the phone will use the session timer even if
the remote party does not support this feature. If it is set to "No", the phone will enable the session timer only when the
Force Timer
remote party supports this feature. To turn off the session timer, select "No".
The default setting is "No".
Force Sets the SIP message type for refresh the session. If it is set to "Yes", the Session Timer will be refreshed by using the SIP
INVITE INVITE message. Otherwise, the phone will use the SIP UPDATE or SIP OPTIONS message. Default is "No".

Account/Codec Settings

Preferred Vocoder
Codec Negotiation Priority
Use First Matching
Vocoder in 200OK SDP
iLBC Frame Size
Opus Payload Type
DTMF
Table 13: Account/SIP Settings
Preferred Vocoder
Lists the available and enabled Audio codecs for this account. Users can enable the specific audio codecs by
moving them to the selected box and set them with a priority order from top to bottom. This configuration
will be included with the same preference order in the SIP SDP message. The Supported vocoders are:
PCMU, PCMA, G722, G729A/B, iLBC and Opus.
Configures the phone to use which codec sequence to negotiate as the callee. When set to "Caller", the
phone negotiates by SDP codec sequence from received SIP Invite; When set to "Callee", the phone
negotiates by audio codec sequence on the phone.
Includes only the first matching vocoder in its 200OK response, otherwise it will include all matching
vocoders in same order received in INVITE.
The default setting is No.
Specifies iLBC packet frame size (20ms or 30ms). Default is 30ms.
Determines payload type for Opus codec. The valid range is between 96 and 126.
Default is 123.
Specifies the mechanism to transmit DTMF (Dual Tone Multi-Frequency) signals. There are 3 supported
modes:
● In audio, which means DTMF is combined in the audio signal (not very reliable with low-bit-rate
codecs);
● RFC2833, which means to specify DTMF with RTP packet. Users could know the packet is DTMF in
the RTP header as well as the type of DTMF.
● SIP INFO, which uses SIP INFO to carry DTMF. The defect of this mode is that it is easily to cause
desynchronized of DTMF and media packet if the SIP and RTP messages are required to transmitted,