Aastra Telecom 41-001343-02 IP Phone User Manual


 
Operational, Basic, and Advanced Parameters
A-99 41-001343-02 REV04 – 05.2014
RTP, Codec, DTMF Global Settings
Global Settings
Parameter
sip rtp port
Configuration Files
aastra.cfg, <model>.cfg, <mac>.cfg
Description Indicates the port through which the RTP packets are sent. This value must specify the
beginning of the RTP port range on the gateway or router.
The RTP port is used for sending DTMF tones and for the audio stream. Your network
administrator may close some ports for security reasons. You may want to use this
parameter to send RTP data using a different port.
Note:
The phones support decoding and playing out DTMF tones sent in SIP INFO requests.
The following DTMF tones are supported:
Support signals 0-9, #, *
Support durations up to 5 seconds
Format Integer
Default Value 3000
Range Not Applicable
Example sip rtp port: 3000
Parameter
sip use basic codecs
Configuration Files
aastra.cfg, <model>.cfg, <mac>.cfg
Description Enables or disables basic codecs (G.711 u-Law, G.711 a-Law, G.729). Enabling this
parameter allows the IP phone to use the basic Codecs when sending/receiving RTP
packets.
Format Boolean
Default Value 0
Range 0 - Disable
1 - Enable
Example sip use basic codecs: 1
Parameter
sip amr codec payload format
Configuration Files
aastra.cfg, <model>.cfg <mac>.cfg
Description Specifies the payload format for the AMR/AMR-WB codec.
Format Integer
Default Value 0
Range 0-2
0 (Enable bandwidth efficient mode per RFC, no octet-aligned header is in the INVITE
SDP [default]).
1 (Enable octet-aligned mode and add octet-align:1 in SDP,
negotiate mode for incoming calls).
2 (Disable octet-aligned mode and add octet-align:0 in SDP,
negotiate mode for incoming calls).
Example sip amr codec payload format: 1