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This feature module describes the Cisco IOS feature, UNI-OSP, which is first introduced in Cisco IOS Release 12.1(2)T. The UNI-OSP feature allows the authentication of outgoing Voice over IP (VoIP) telephone connections, using the Open Settlement Protocol (OSP). It includes information on the benefits of this new feature, supported platforms, and configuration examples. This document includes the following sections:
The UNI-OSP feature allows the Cisco AS5300 universal access server gateway to control access to the PSTN from a VoIP network. Authentication of VoIP calls to the PSTN allows customers to limit calls to authorized users and to prevent unauthorized usage of limited telephony resources. This feature can also be used as part of the enabling technology required to implement a "click-to-talk" feature. Click-to-talk allows users to initiate a VoIP telephone connection from a web server page.
The basis for this feature is the Open Settlement Protocol (OSP), which is the standard Cisco settlement protocol. OSP is a protocol based on Secure Sockets Layer (SSL), which authenticates an IP session and authorizes the usage of network resources. Open Settlement Protocol uses a combination of Hypertext Transfer Protocol (HTTP), Extensible Markup Language, and Secure Sockets Layer (SSL 3.0) to perform transfer pricing, authorization, and to indicate usage information.
The OSP implementation allows two Cisco AS5300 gateways to use OSP to authorize and bill PSTN calls routed over an IP network. With the Settlement feature, calls always originate in the PSTN network, are authorized on an incoming gateway, and carry secure token information to an outgoing gateway. The UNI-OSP feature allows a single Cisco AS5300 gateway to use OSP to authenticate VoIP calls to the PSTN.
The UNI-OSP feature uses the Open Settlement Protocol (OSP) to provide the following benefits to users of the Cisco AS5300 gateway:
Before the introduction of the UNI-OSP feature, the Cisco AS5300 gateway supported OSP with two gateways, and allowed authentication of connections originating in the PSTN, to make connections over a VoIP network. The UNI-OSP feature allows customers to authenticate connections originating in a VoIP network with destinations in the PSTN.
One application for UNI-OSP is to implement a click-to-talk functionality, Click-to-talk allows a customer or other user browsing a web page to click on a link and to initiate a telephone connection to the appropriate representative. The following subsections describe these benefits in greater detail, and explain the supporting infrastructure required for implementation.
The UNI-OSP feature, illustrated in Figure 1, allows a single Cisco AS5300 gateway to use OSP to authenticate VoIP calls to the PSTN.
To implement VoIP authentication, the Cisco AS5300 must be connected to the Internet or intranet and to an OSP server. The OSP server, which can be any properly configured Windows NT or UNIX server, communicates over a Computer Telephony Interface (CTI) link to a Class 5 switch or PBX.
When a VoIP device sends an H.323 setup message to the Cisco AS5300, the destination number (DNIS) of the call is matched to a POTS dial peer configured on the Cisco AS5300 gateway. The Cisco AS5300 then sends an OSP authorization request, containing the call ID (a 16-bit unique number), and a calling and called number (E.164 ANI/DNIS).
The OSP server sends an authorization response back to the Cisco AS5300, which then initiates a call to a PBX or Class 5 switch over one of its T1-PRI spans. When the Cisco AS5300 detects that a call has ended, it transmits usage information to the OSP server, informing it that the call has terminated.
Note that the "src-info" field of the OSP authorization request contains the IP address of the caller's PC, with all "." characters removed, and each segment right justified. For example, the ANI field for a call originating from IP address 171.69.221.2 would appear as "172069221002".
As illustrated in Figure 2, the UNI-OSP feature can also be used when implementing a "click-to-talk" function on web server pages.
When using the click-to-talk function, a customer selects a link on a web page indicating that they would like to talk to a customer or technical support representative. The web server then launches a pre-installed soft-phone on the web browser machine through a browser plug-in. The web server supplies the PC softphone application with the telephone destination number (DNIS) of the appropriate agent, and the route point, queue, and IP address of the Cisco AS5300.
When the softphone sends an H.323 setup message to the Cisco AS5300, the destination number (DNIS) is matched to a POTS dial peer configured on the Cisco AS5300 gateway. An authorization request is then sent over the OSP link to the OSP server, containing the call ID (a 16 -bit unique number), and a calling and called number (E.164 ANI/DNIS). Because the originating device is a PC soft-phone, the ANI field contains the IP address of the PC.
The OSP server compares the IP address received in the ANI field with those clients that have pressed the Click-To-Talk link. The OSP server sends an authorization response to the Cisco AS5300, containing the E.164 number of the appropriate agent, based on the web page from which Click-To-Talk was initiated.
When the Cisco AS5300 initiates a call to a PBX or Class 5 switch, the arriving call causes a setup indication to appear on the switch or PBX. The CTI link between the PBX or switch and the OSP server informs the OSP server of the incoming call, and includes information such as the DNIS, ANI (IP address of the caller's PC), and the incoming trunk line. The OSP server then has sufficient information to route the call to the appropriate customer or technical support agent queue.
Many customers have firewalls that do not permit H.323 communications with endpoints outside the enterprise. These customers may open static firewall access to UDP ports 1720 and 16384 through 32767 to enable communications with public Internet endpoints. However, Cisco's PIX firewall (as of Cisco IOS Release 12.0(3)T) allows temporary access to be established when an outbound H.323 setup message is detected, based on Context-Based Access Control (CBAC). This eliminates the need to open static holes in the firewall.
The Settlement for Packet Telephony feature is dependent upon the interoperability of the following feature:
Cisco Customer Documentation:
Other Documentation:
Ensure that your Cisco AS5300 has the following memory requirement:
16 MB Flash and 64 MB DRAM memory minimum.
To configure the Cisco AS5300, perform the following tasks:
The following table shows the commands required to configure the Cisco AS5300 router for the UNI-OSP feature.
| Command | Purpose | |
|---|---|---|
Step 1 | Router# configure terminal | Enters the global configuration mode. |
Step 2 | Router(config)# settlement number | Enters the Settlement mode and configure the Settlement provider number. The settlement command puts you in the Settlement command mode. |
Step 3 | Router(config-settlement# type osp | Configures the Settlement provider type. In Cisco IOS Release 12.0(4)XH, OSP is the only type available. |
Step 4 | Router(config-settlement# url <server url> | This step can be repeated if the settlement provider has more than one service point. |
Step 5 | Router(config-settlement)# no shutdown | Bring up the settlement provider. |
The following table shows the commands required to configure the POTS dial-peer for using the UNI-OSP feature on an Cisco AS5300 router:
| Command | Purpose | |
|---|---|---|
Step 1 | Router(config-settlement)# dial-peer voice number pots | Enters the dial-peer configuration mode to configure a POTS dial-peer. |
Step 2 | Router(config-settlement)# destination-pattern [+]string[T] | Configures the dial peer's destination pattern. Enters the number or pattern of the outbound called number. · The plus symbol (+) can be used to indicate an E.164 standard number. |
Step 3 Step 3 | Router(config-settlement)# application session | Configures the "application session" attribute for the POTS dialpeer. |
Step 4 | Router (config-settlement) # port port-number | Associates the POTS dial peer with a specific voice port. |
Use the show settlement command to verify your configuration. See Figure 3.
For UNI-OSP settlement, configure the URL of the settlement server and indicate that the settlement type is issuing-osp, as in the following example.
Router# settlement 0 Router# type uni-osp Router# url 172.100.100.1
For the POTS dial peer, configure it like the example below. This example shows how to configure a destination number of 1000, where 1000 is the route point/DNIS of a device connected to the PBX to which the T1 line is attached:
dial-peer voice 111 pots destination pattern 1000 application session port 0:D session target settlement:0 dial-peer voice 222 voip incoming called-number 1000 codec g711ulaw application session
When the H.323 call "setup" message arrives, it should have the destination number (DNIS) of "1000". The Cisco AS5300 will then handle it as a settlement call, and route an authorization request to the OSP server.
The document Settlement Plus Roaming and PKI Multiple Roots on Cisco Access Platforms documents the full command set required for configuring the UNI-OSP feature. This document is available at the following URL:
The only modification to the Settlement commands that is introduced for the UNI-OSP feature is a new parameter for the type command, which is documented in this section.
To point to the provider type and the specific Settlement server, enter the type command in the Settlement configuration mode. This command line defines the Settlement server that is doing the accounting, and enables the server to do the accounting.
In Cisco IOS Release 12.1(1)T, the uni-osp Settlement type is introduced to allow authenticating VoIP calls by a Cisco AS5300.
Use the no form of this command to disable this command.
type {osp | uni-osp}
Syntax Description
osp Enables the Open Settlement Protocol server type. uni-osp Enables authentication of VoIP calls to the PSTN, using a single Settlement server.
Defaults
The default is osp.
Command Modes
Settlement configuration
Command History
12.0(4)XH1 This command was introduced. 12.1(1)T The uni-osp settlement type was introduced.
Release
Modification
Examples
settlement 0 type uni-osp
Related Commands
connection-timeout Sets the connection timeout. customer-id Sets the customer identification. device-id Sets the device identification. encryption Specifies the encryption method. max-connection Sets the maximum simultaneous connections. response-timeout Sets the response timeout. retry-delay Sets the retry delay. retry-limit Sets the connection retry limit. session-timeout Sets the session timeout. settlement Enters the Settlement configuration mode. show settlement Displays the configuration for all Settlement server transactions. shutdown/no shutdown Brings up/shuts down the Settlement provider. url Specifies the Internet service provider address.
Command
Description
ANI---automatic number identification. The number of the calling party in SS7 (signaling system 7) communications.
CTI---Computer Telephony Integration. A protocol used to communicate between telephony and computer systems.
DNIS---dialed number information service. The number of the called (destination) party in SS7 (signaling system 7) communications.
E.164---An ITU-T recommendation for international telecommunication addressing, based on traditional telephone numbers. This standard defines source and destination numbers (ANI/DNIS) for a voice connection.
H.323---A series of protocols that define methods for establishing voice and multi-media connections between computer systems over circuit-switched media.
ISP---Internet service provider.
IVR---Interactive Voice Response. A Cisco IOS software voice feature for internet telephony service providers.
OSP---Open Settlement Protocol.
POTS---plain old telephone service. Used to refer to traditional telephony devices and technology.
PSTN---Public Switched Telephone Network.
SSL---Secure Sockets Layer.
TCP---Transmission Control Protocol.
UDP---User Datagram Protocol.
VoIP---Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with POTs-like functionality, reliability, and voice quality. VoIP is a blanket term, which generally refers to Cisco's standards based (for example H.323) approach to IP voice traffic.
Posted: Fri May 26 08:28:26 PDT 2000
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