AudioCodes MediaPack MP-118 User Manual

Version 6.6 471 MP-11x & MP-124
User's Manual 43. Configuration Parameters Reference
Parameter Description
[SIPForceRport]
Determines whether the device sends SIP responses to the UDP port
from where SIP requests are received even if the 'rport' parameter is not
present in the SIP Via header.
 [0] = (Default) Disabled. The device sends the SIP response to the
UDP port defined in the Via header. If the Via header contains the
'rport' parameter, the response is sent to the UDP port from where
the SIP request is received.
 [1] = Enabled. SIP responses are sent to the UDP port from where
SIP requests are received even if the 'rport' parameter is not present
in the Via header.
Web: Reject Cancel after
Connect
CLI: reject-cancel-after-
connect
[RejectCancelAfterConne
ct]
Determines whether the device accepts or rejects a SIP CANCEL
request received after the receipt of a 200 OK, during an established
call.
 [0] = (Default) Accepts the CANCEL, by responding with a 200 OK
and terminating the call session.
 [1] = Rejects the CANCEL, by responding with a SIP 481
Call/Transaction Does Not Exist, and maintaining the call session.
Web: Verify Received
RequestURI
CLI: verify-rcvd-requri
[VerifyReceevedRequest
Uri]
Enables the device to reject SIP requests (such as ACK, BYE, or re-
INVITE) whose user part in the Request-URI is different from the user
part received in the Contact header of the last sent SIP request.
 [0] Disable = (Default) Even if the user is different, the device
accepts the SIP request.
 [1] Enable = If the user is different, the device rejects t
he SIP request
(BYE is responded with 481; re-INVITE is responded with 404; ACK
is ignored).
Web: Max Number of
Active Calls
EMS: Maximum
Concurrent Calls
[MaxActiveCalls]
Defines the maximum number of simultaneous active calls supported by
the device. If
the maximum number of calls is reached, new calls are not
established.
The valid range is 1 to the maximum number of supported channels.
The default is the maximum available channels (i.e., no restriction on
the maximum number of calls).
Web: QoS statistics in SIP
Release Call
[QoSStatistics]
Enables the device to include call quality of service (QoS) statistics in
SIP BYE and SIP 200 OK response to BYE, using the proprietary SIP
header X-RTP-Stat.
 [0] = Disable (default)
 [1] = Enable
The X-RTP-Stat header provides the following statistics:
 Number of received and sent voice packets
 Number of received and sent voice octets
 Received packet loss, jitter (in ms), and latency (in ms)
The X-RTP-Stat header contains the following fields:
 PS=<voice packets sent>
 OS=<voice octets sent>
 PR=<voice packets received>
 OR=<voice octets received>
 PL=<receive packet loss>
 JI=<jitter in ms>
 LA=<latency in ms>
Below is an example of the X-RTP-Stat header in a SIP BYE message:

AudioCodes MediaPack MP-118 Specifications

General

Category	Gateway
Analog Ports	8 FXS ports
Voice Codecs	G.711, G.723.1, G.726, G.729A/B
Operating Temperature	0°C to 40°C (32°F to 104°F)
Storage Temperature	-20°C to 70°C (-4°F to 158°F)
Protocols	SIP, MGCP, H.323
Model	AudioCodes MediaPack MP-118
Fax Support	T.38
Management	Web-based, SNMP, CLI
Security	SRTP, TLS, HTTPS
Power Supply	AC 100-240V, 50-60Hz
Humidity	10% to 90%, non-condensing
Network Interfaces	2 x 10/100Base-T Ethernet