User's Manual
Table Of Contents
- Cisco WRP500 Administration Guide
- Contents
- 1
- WRP500 Features and Benefits
- Deployment Models
- Local Area Network Guidelines
- Special Requirements for Voice Deployments
- WRP500 Maintenance Operations
- Remote Provisioning
- Configure NAT Mapping
- Configure NAT Mapping with a Static IP Address
- Step 1 Log in as administrator.
- Step 2 Under the Voice menu, click SIP.
- Step 3 In the NAT Support Parameters section, enter the following settings:
- Step 4 Under the Voice menu, click Line 1 or Line 2 to choose the line interface that you want to modify.
- Step 5 In the NAT Settings section, enter the following settings:
- Step 6 Click Save Settings.
- Configure NAT Mapping with STUN
- Step 1 Log in as administrator.
- Step 2 Under the Voice menu, click SIP.
- Step 3 In the NAT Support Parameters section, enter the following settings:
- Step 4 Under the Voice menu, click Line 1 or Line 2 to choose the line interface that you want to modify.
- Step 5 In the NAT Settings section, enter the following settings:
- Step 6 Click Save Settings.
- Determine Whether the Router Uses Symmetric or Asymmetric NAT
- Step 1 Make sure you do not have firewall running on your computer that could block the syslog port (port 514 by default).
- Step 2 Log in as administrator.
- Step 3 To enable debugging, complete the following tasks:
- Step 4 To collect information about the type of NAT your router is using, complete the following tasks:
- Step 5 To enable SIP signaling, complete the following task:
- Step 6 Click Submit.
- Step 7 View the syslog messages to determine whether your network uses symmetric NAT. Look for a warning header in the REGISTER messages, such as Warning: 399 spa "Full Cone NAT Detected.”
- Configure NAT Mapping with a Static IP Address
- Firewalls and SIP
- Configure SIP Timer Values
- Analog Telephone Adapter Operations
- ATA Software Features
- Register to the Service Provider
- Step 1 Log in as administrator.
- Step 2 Under the Voice menu, click Line 1 or Line 2 to choose the line interface that you want to modify.
- Step 3 In the Proxy and Registration section, enter the Proxy.
- Step 4 In the Subscriber Information section, enter the User ID and Password.
- Step 5 Click Submit. The devices reboot.
- Step 6 To verify your progress, perform the following tasks:
- Manage Caller ID Service
- Optimize Fax Completion Rates
- Step 1 Ensure that you have enough bandwidth for the uplink and the downlink.
- Step 2 To optimize G.711 fallback fax completion rates, set the following on the Line tab of your ATA device:
- Step 3 If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax relay) and enable fax using modem passthrough.
- Step 4 Enable T.38 fax on the WRP500 by configuring the following parameter on the Line tab for the FXS port to which the FAX machine is connected:
- Step 5 If you are using a Cisco media gateway use the following settings:
- Fax Troubleshooting
- Step 1 Verify that your fax machine is set to a speed between 7200 and 14400.
- Step 2 Send a test fax in a controlled environment between two ATAs.
- Step 3 Determine the success rate.
- Step 4 Monitor the network and record the following statistics:
- Step 5 If faxes fail consistently, capture a copy of the voice settings by selecting Save As > Web page, complete from the administration web server page. You can send this configuration file to Technical Support.
- Step 7 Identify the type of fax machine connected to the ATA device.
- Step 8 Contact technical support:
- Silence Suppression and Comfort Noise Generation
- Configure Dial Plans
- About Dial Plans
- Edit Dial Plans
- Enter the Line Interface Dial Plan
- Step 1 Start Internet Explorer, connect to the Configuration Utility, choose Voice > Admin Login. If prompted, enter the administrative login provided by the Service Provider. (The default username and password are both admin.)rovided by your Service...
- Step 2 Under the Voice menu, click Line 1 or Line 2, depending on the line interface that you want to configure.
- Step 3 Scroll down to the Dial Plan section.
- Step 4 Enter the digit sequences in the Dial Plan field. For more information, see “About Dial Plans,” on page 11.
- Step 5 Click Submit.
- Reset the Control Timers
- Step 1 Start Internet Explorer, connect to the Configuration Utility, choose Voice > Admin Login. If prompted, enter the administrative login provided by the Service Provider. (The default username and password are both admin.)rovided by your Service...
- Step 2 Under the Voice menu, click Regional.
- Step 3 Scroll down to the Control Timer Values section.
- Step 4 Enter the desired values in the Interdigit Long Timer field and the Interdigit Short Timer field. Refer to the definitions at the beginning of this section.
- Enter the Line Interface Dial Plan
- Secure Call Implementation
- Info page
- System page
- SIP page
- Regional page
- Line page
- Line Enable section
- Streaming Audio Server (SAS) section
- NAT Settings section
- Network Settings section
- SIP Settings section
- Proxy and Registration section
- Subscriber Information section
- Supplementary Service Subscription section
- Audio Configuration section
- Dial Plan section
- FXS Port Polarity Configuration section
- User page
- Setup
- Wireless Configuration
- Security
- Access Restrictions
- Applications and Gaming
- Administration
- Status
REVIEW DRAFT #1—CISCO CONFIDENTIAL
A-30
Cisco WRP500 Administration Guide
Appendix A Advanced Voice Fields
Line page
Voice tab > Line page >
NAT Settings section
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 value is
used by the 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 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 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.
NAT Keep Alive Dest
Destination that should receive NAT keep alive messages. If the
value is $PROXY, the messages are sent to the current proxy server
or outbound proxy server.
The default is $PROXY.