AudioCodes 400HD Series - Configuring Media Bypass; Configuring RTCP Extended Report; Table 3-37: RTCP_XR Parameter

400HD IP Phones Series - Teams Compatible
Administrator's Manual 84 Document #: LTRT-09960
3.4.16.4 Configuring RTCP Extended Report
This section shows how to configure Extended Report for RTP Control Protocol (RTCP-XR)
working mode.
 To configure RTCP_XR:
• Open the Configuration File page (Management tab > Manual Update >
Configuration File) and configure the parameter using the table below as
reference.
Table 3-37: RTCP_XR Parameter
Parameter Description
voip/rtcp_xr/vq_statistics/mode
Sets RTCP_XR working mode.
Select either:
 [DISABLE] (default). In this state, no RTCP-XR
events are retrieved from the phone and the SIP
PUBLISH is not sent, regardless of the state of
parameter 'qoe_publish_enabled' (see below).
 [EVENTS_ONLY]. In this state, RTCP-XR events
with voice quality parameter calculations are sent
internally on the phone every five seconds. Each
calculation is made on the basis of these RFC 3611
parameters: BT=7, block length = 8SSRC of
source, loss rate, discard rate, burst density, gap
density, burst duration, gap duration, round trip
delay, end system delay, signal level, noise level,
Gmin, R factor, ext. R factor, MOS-LQ, MOS-CQ,
RX config, JB nominal, JB maximum and JB abs
max. The phone sends the summarized RTCP-XR
events to the Skype for Business server / EMS via
SIP SERVICE messages (in Genesis-SIP, SIP
PUBLISH messages are used).
 [REMOTE_AND_EVENTS]. In this state, the phone
sends RTCP-XR events to the remote calling party
(i.e. party A sends these events to party B) every
five seconds during the VoIP session. The phone
sends the summarized RTCP-XR events to the
Skype for Business server / EMS via SIP SERVICE
messages (in Genesis-SIP, SIP PUBLISH
messages are used).
3.4.16.5 Configuring Media Bypass
Media bypass allows a phone to send media directly to the SBC or PSTN gateway,
eliminating the Cloud Connector Edition (CCE) from the media path when possible, thereby
reducing latency, the possibility of packet loss and the number of points of potential failure,
and thereby improving voice quality.
The feature is only applicable:
 to Skype for Business online phones with one or more CCEs interconnected to SBCs
or to gateways
 if enabled by inband provisioning parameter
 if the phone receives a valid bypassID from a CCE Web service
 to phones connected to CCE over an internal IP network [External phones do not have