Use the SIP Settings Page

Using the SIP Settings

This section outlines how to use the SIP Settings page.

To use the SIP Settings Page

1. Choose VoIP > SIP.
   
   

2. Configure settings using the information in the following table as a guide. When you have finished configuring settings, click Submit to make your changes take effect.

   SIP Settings Parameters

   Item	Description
   SIP Protocol Settings Configures SIP ALG settings. Refer to Configure SIP Protocol Settings.
   SIP Server Address	Specifies the SIP server IP address or URL.
   SIP Server Port	Specifies the SIP server port number.
   SIP Server Transport	Specifies SIP server transport: • Pass Through • UDP • TCP • TLS If the Client/Phone transport is different from the Server transport, the system does the transport conversion. The system cannot convert from UDP to TCP or TLS while in ALG mode.
   Use Custom Domain	Configures a custom domain name other than the default SIP domain, which is the same as the SIP Sever address.
   SIP Server Domain	Specifies the domain name of the SIP server that accepts forwarded client traffic.
   List of SIP Servers The system supports auto failover to secondary SIP servers, if it is provided with a list of SIP servers. There are two methods to obtain the list of SIP servers: • DNS SRV Lookup on SIP Server URL—Default; done automatically if the address of the SIP Server is a URL. • Manually enter the list—The Create button opens the List of SIP Servers window. Refer to Manage the List of SIP Servers.
   Add a new proxy Specifies the IP address and port of the SIP Server or Session Border Controller (SBC).
   Enable Multi-homed Outbound Proxy Mode	Allows phones behind the same system to use the default soft-switch or a specified soft-switch.
   Enable Transparent Proxy Mode	Allows the system to intercept SIP messages from a LAN-side phone regardless of the Outbound Proxy and SIP Proxy values configured in the phone.
   Limit Outbound to listed SIP Servers	Restricts the number of permitted proxies that are specified in the Allowed SIP Proxies window. Limit Outbound is the default in Transparent Proxy Mode. The system only process outbound messages intended for a pre-configured list of proxies and SIP servers. Un-checking this box allows the system to process all outbound messages for any SIP proxy/server.
   Limit Inbound to listed SIP Servers	Allows the system to process only inbound messages from a pre-configured list of proxies and SIP servers. Un-checking the box allows the system to process all inbound messages from any SIP proxy/server.
   Dynamic List of SIP Proxies Support	Allows creation of a dynamic list of SIP proxies. Enabling this option will cause the system to forward all REGISTER requests to WAN side SIP server addresses that are not pre-configured in the list of SIP servers or Allowed Proxies.
   Include UPDATE In Allow	This option is applicable for B2BUA calls only. Enabling this option will add the UPDATE method in the Allow header if it is not already present in the incoming request. Disabling this option will forward Allow header as in the original request.
   PRACK Support	Enabling or disabling this option provides SIP PRACK interoperability support. SIP PRACK interoperability provides B2BUA support to allow an endpoint that does not support PRACK to inter-operate with an endpoint that requires PRACK support.
   GEOLOCATION Support	Enabling or disabling this option provides SIP Geolocation header support. When it is enabled, the system parses the header and passes it to the remote side. When it is disabled, the system skips the header.
   Call Audit Support	This option is applicable for B2BUA calls only. Enabling this option will result in all OPTIONS and UPDATE messages without an SDP, received within an existing dialog, to receive a 200 OK response from Unable to show "metadata-from": No such page "_space_variables" and all OPTIONS/UPDATE messages with SDP to be forwarded to the other end.
   Enable PAssociated-URI Support	Enabling this option allows inbound calls to any of the P-Associated URI of a hosted client. The PAssociated URI is given by the Registrar and is kept in  Unable to show "metadata-from": No such page "_space_variables" for processing the signaling packets of the incoming call.
   SIP Use New Port on Hold Resume	Enable this option if you want SIP to use a new port on hold-resume.
   Stale client time (m)	Automatically deletes SIP clients that have not registered within the given time period. If no REGISTER requests have been received from the client during this time, the client will be automatically deleted. Default is 1440 minutes. It is recommended that this time be much longer than the normal re-registration period in order to avoid clients being accidentally deleted. Refer to Configure the SIP Stale Timer.
   List of Allowed [Maximum 50] SIP Servers Lists the SIP servers or registrars that are allowed when enabling the Limit Outbound (for transparent mode only) and Limit Inbound (for transparent as well as non-transparent mode) options. The SIP server address is always included and does not have to be in this list. Lists the IP addresses of outbound proxies or registrars that are allowed in transparent proxy mode. The List of Allowed [Maximum 50] SIP Servers table. Refer to Configure Allowed SIP Servers.
   SDP Modifications Modifies codecs in the SDP. Refer to Configure SDP Modifications Settings.
   SDP Codec Operation	Selects the type of operation to be performed on codec payloads in the SDP. The following options are available: • No Action • Delete given codecs—The list of codecs given in the codecs list will be deleted from the SDP. • Only Allow given codecs—If the SDP has any codecs not listed in the list of codecs then it will be removed from the SDP. • Change Codec Order—If the codecs given in the codecs list occur in the SDP, the system ensures that the list is ordered entered.
   SDP Section that will be modified	Selects the media section for searching the codecs: • Audio—Search for audio codecs. • Video—Search for video codecs.
   Codecs	Specifies a comma separated list of codecs to search for. The codecs should be referred by their names such as PCMU, PCMA, G723, G729, H261, and so on. Refer to Real-Time Transport Protocol (RTP) Parameters at the following link for standardized codec names: https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml.
   Reject when No Match Codec	Select the check-box to allow the Unable to show "metadata-from": No such page "_space_variables" to reject the incoming INVITE with a 488 SIP response code when no remaining codec is left in the SDP after codec operation. When the box is unchecked, the Unable to show "metadata-from": No such page "_space_variables" continues forwarding the SDP after codec operation to the remote side. This option is only available for SIP B2BUA mode.
   Strip Matched Expressions	Enter expression to match in the configured SDP media session. Once matched, the matched line or text will be removed from SDP. If exact text match is used, the space needs to be exact same.
   Priority Numbers The Unable to show "metadata-from": No such page "_space_variables" allows you to set up to 4 priority numbers for calls that will take precedence over non-priority numbers. These priority numbers are treated for high priority call completion. For example, a company might have priority numbers for special applications assigned to speed-dial number keys that have been programmed into phone for emergency or security. These priority numbers are configured in the Unable to show "metadata-from": No such page "_space_variables" through the following interface. When a priority number is called and the call limit has been reached (via the license key limit or CAC), the Unable to show "metadata-from": No such page "_space_variables" will drop the non-priority calls and complete the call that has been assigned the priority number. Calls with priority numbers cannot be dropped if all active calls are assigned priority numbers and additional call attempts are also made with a priority number.
   Priority Number 1-4	Enter a number in the Priority Number field, for example, “777.”

Configuring SRTP

Secure Real Time Protocol (SRTP) support for Transport Layer Protocol (TLS) hosted clients requires an outbound rule for routing the calls through B2BUA if the clients use TLS/SRTP for signaling and media respectively. TLS/SRTP inbound calls can now be routed through B2BUA without writing any match rules.

Outgoing REGISTER messages are routed from hosted clients through B2BUA. This is necessary for successful registration when the LAN side client and WAN side softswitch use different protocols.

Supported Platforms:

To Enable the SRTP

1. Choose VoIP from the Configuration Menu.
   
   

2. Select the Route all SIP signalling through B2BUA checkbox.

3. Click Submit to save your changes.

You can also enable the feature using CLI by entering ALG_FORCE_B2BUA_ROUTE=on (off) in the /etc/config/alg_defs.conf file.

Feature Limitations

• Outbound calls from hosted clients still require the single B2BUA outbound rule.
• Even when the feature is enabled, local calls from hosted clients with RTP to a B2BUA Trunking Device using SRTP will have white noise if the outbound calls from hosted clients are not routed using a B2BUA rule. To have this scenario working, you must use a B2BUA outbound rule.
• TLS/SRTP match rule for B2BUA is supported on the
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  7000 Series.
• The ALG client list can now be added as the match rule for the B2BUA. This allows for the devices to register from the LAN and build the client list. Once the list is built, all inbound and outbound calls process through the B2BUA, giving a seamless functionality for WAN side TLS/SRTP.