Administrator's Manual
4. Configuration File-based Management
Version 1.4
73
July 2010
4.2.5.4 Media Streaming Parameters
Table
4-16: Media Streaming Configuration File Parameters
Parameter
Description
voip/media/dtmf_payloa
d
Defines the RTP payload type used for RFC 2833 DTMF relay packets.
The valid range is 96 to 127. The default value is 101.
voip/media/media_port
Defines the starting port range for Real Time Protocol (RTP) voice transport.
The valid range is 1024 to 65535. The default value is 4000.
voip/media/rtp_mute_on
_hold
Mute sending RTP packets to remote in HOLD state.
[0]
- Disabled. RTP packets are sent to remote end when in HOLD state.
[1]
(default) - Enabled. RTP packets are not sent to remote end when in
HOLD state.
voip/media/out_of_band
_dtmf
DTMF transport mode.
[
INBAND]
Inband
[
RFC2833]
RFC 2833 (default)
[VIA_SIP]
Via SIP
Quality of Service (QoS)
voip/media/media_tos
QoS in hexadecimal format. This is a part of the IP header that defines the
type of routing service to tag outgoing voice packets originated from the
phone. It informs routers that this packet must receive a specific QoS. The
default value is
0xb8
.
Values can be set in decimal (e.g. 184) or hexadecimal (e.g. 0xb8).
4.2.5.5 SIP Signaling Parameters
4.2.5.5.1 General Parameters
Table
4-17: SIP General Configuration File Parameters
Parameter
Description
voip/signalling/sip/trans
port_protocol
Determines the transport layer for outgoing SIP calls initiated by the phone.
[UDP]
UDP (default)
[TCP]
TCP
[TLS]
TLS
voip/signalling/sip/port
Defines the local SIP port (UDP, TCP or TLS) for SIP messages.
The valid range is 1024 to 65535. The default value is 5060.
voip/signalling/sip/proxy
_gateway
Assigns a name to the phone. The name is used as the host part of the SIP
URI in the From header.
Notes:
Ensure that the name you choose is the one with which the Proxy is
configured to identify the phone.
If not specified, the phone's IP address is used (default).
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