Cisco SPA1001 Administration Guide - Page 118

NAT Settings

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Line Tab SAS Enable SAS DLG Refresh Intvl SAS Inbound RTP Sink Chapter 5 Linksys ATA Field Reference To enable the use of the line as a streaming audio source, select yes. Otherwise, select no. If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming calls and streams audio RTP packets to the caller. The default is no. If this is not zero, it is the interval at which the streaming audio server sends out session refresh (SIP re-INVITE) messages to determine whether the connection to the caller is still active. If the caller does not respond to the refresh message, the Linksys IP phone ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled). The default is 30. This setting works around devices that do not play inbound RTP if the streaming audio server line declares itself as a send-only device and tells the client not to stream out audio. Enter a Fully Qualified Domain Name (FQDN) or IP address of an RTP sink; this is used by the Linksys IP phone's streaming audio server line in the SDP of its 200 response to an inbound INVITE message from a client. The purpose of this parameter is to work around devices that do not play inbound RTP if the SAS line declares itself as a send-only device and tells the client not to stream out audio. This parameter is a FQDN or IP address of a RTP sink to be used by the SPA SAS line in the SDP of its 200 response to inbound INVITE from a client. It will appear in the c = line and the port number and, if specified, in the m = line of the SDP. If this value is not specified or equal to 0, then c = 0.0.0.0 and a=sendonly will be used in the SDP to tell the SAS client to not to send any RTP to this SAS line. If a non-zero value is specified, then a=sendrecv and the SAS client will stream audio to the given address. Special case: If the value is $IP, then the SAS line's own IP address is used in the c = line and a=sendrecv. In that case the SAS client will stream RTP packets to the SAS line. The default value is empty. NAT Settings Field NAT Mapping Enable NAT Keep Alive Enable NAT Keep Alive Msg Description To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. The default is no. To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. The default is no. Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent. The default is $NOTIFY. 5-34 Linksys IP Phone Administrator Guide Document Version 3.2

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5-34
Linksys IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
Line Tab
NAT Settings
SAS Enable
To enable the use of the line as a streaming audio source, select yes. Otherwise, select no.
If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming
calls and streams audio RTP packets to the caller.
The default is no.
SAS DLG Refresh Intvl
If this is not zero, it is the interval at which the streaming audio server sends out session
refresh (SIP re-INVITE) messages to determine whether the connection to the caller is
still active. If the caller does not respond to the refresh message, the Linksys IP phone
ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the
session refresh is disabled).
The default is 30.
SAS Inbound RTP Sink
This setting works around devices that do not play inbound RTP if the streaming audio
server line declares itself as a send-only device and tells the client not to stream out audio.
Enter a Fully Qualified Domain Name (FQDN) or IP address of an RTP sink; this is used
by the Linksys IP phone’s streaming audio server line in the SDP of its 200 response to
an inbound INVITE message from a client.
The purpose of this parameter is to work around devices that do not play inbound RTP if
the SAS line declares itself as a send-only device and tells the client not to stream out
audio. This parameter is a FQDN or IP address of a RTP sink to be used by the SPA SAS
line in the SDP of its 200 response to inbound INVITE from a client. It will appear in the
c = line and the port number and, if specified, in the m = line of the SDP.
If this value is
not specified or equal to 0, then c = 0.0.0.0 and a=sendonly will be used in the SDP to tell
the SAS client to not to send any RTP to this SAS line. If a non-zero value is specified,
then a=sendrecv and the SAS client will stream audio to the given address. Special case:
If the value is $IP, then the SAS line’s own IP address is used in the c = line and
a=sendrecv. In that case the SAS client will stream RTP packets to the SAS line.
The default value is empty.
Field
Description
NAT Mapping Enable
To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes.
Otherwise, select no.
The default is no.
NAT Keep Alive Enable
To send the configured NAT keep alive message periodically, select yes. Otherwise, select
no.
The default is no.
NAT Keep Alive Msg
Enter the keep alive message that should be sent periodically to maintain the current
NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is
$REGISTER, a REGISTER message without contact is sent.
The default is $NOTIFY.