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This feature module describes the Cisco H.323 Voice over IP (VoIP) architecture combined with SS7 signaling functionality for the Cisco AS5300. It includes information on the benefits of the new feature, supported platforms, related documents, and more.
This document includes the following sections:
The Cisco H.323 VoIP with SS7 feature provides SS7 connectivity for voice gateways by using the signaling controller as a protocol translator to control the gateway using the ISDN Q.931 protocol. This feature interacts over the IP network with other Cisco H.323 VoIP with SS7 solutions functioning as an end-to-end SS7 voice network. In addition, it can communicate with other H.323 endpoints, including gateways using non-SS7 signaling such as ISDN PRI and channelized T1.
The access gateway terminates the PSTN trunks, also referred to as bearer channels, that carry the call traffic. The PSTN trunks terminate into either T1 or E1 PRI interfaces on the gateway. In addition, the access gateway performs call control (including originating and terminating call processing/signaling). The gateway referred to in this document is the Cisco AS5300.
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Note Some product labels and packaging might use the term telephony controller. Any references to the telephony controller apply to the signaling controller. |
The following enhancements to dial peer configuration lower complexity of dial planning and reduces the amount of effort in creating dial peer entries:
These new dial string symbols are added: Percent, plus, question mark, period, brackets, and parenthesis.
| 1The "+" symbol can be part of dialing numbers in some countries, where "+" is always be a leading digit in the dialed number. However, this does not conflict with the regular expression rule; "+" in regular expressions will never be a leading symbol. |
When configuring your dial peers, you are provided with an option called the translation rule. This rule applies a translation rule to a calling party number [Automatic Number Identification (ANI)] or a called party number [Dial Number Information Service (DNIS)] for both incoming and outgoing calls within Cisco H.323 voice-enabled gateways. Also, the rule allows translation of the type of number. Refer to the Q.931 ITU specification for details.
To match on a number type for a dial peer call leg, the numbering-type command is used in dial-peer configuration mode.
When a called number is received and matched to a POTS dial peer, the matched digits are stripped and the remaining digits are forwarded to the voice interface. A new command called digit strip makes this default behavior an option.
Currently, dial peer configuration needs multiple dial peers to support a dialing plan. This feature reduces the amount of effort in producing dial peer entries, improves VoIP system performance significantly because of less dial peer search, and uses less memory.
When a called number is received and matched to a POTS dial peer, the matched digits are stripped and the remaining digits are forwarded to the voice interface. A new command called the digit strip makes this default behavior an option. This means you can easily get caller ID and restriction information, and you also don't have to make long-distance calls between small, neighboring countries.
Configuring the Cisco AS5300 for Dial Access and Voice-over-IP
Cisco H.323 VoIP with SS7 Configuration Options and Components
This feature is supported on the following platform:
Cisco IOS software is packaged in feature sets that support specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.
Cisco Feature Navigator is a web-based tool that enables you to determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.
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Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
Platform support for particular Cisco IOS software releases is dependent on the availability of the software images for those platforms. Software images for some platforms may be deferred, delayed, or changed without prior notice. For updated information about platform support and availability of software images for each Cisco IOS software release, refer to the online release notes or, if supported, Cisco Feature Navigator.
No new or modified standards are supported by this feature.
No new or modified MIBs are supported by this feature.
For descriptions of supported MIBs and how to use MIBs, see the Cisco MIB web site on CCO at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
No new or modified RFCs are supported by this feature.
Before you can configure your platform to serve as an H.323 VoIP gateway, you must first:
To configure a Cisco AS5300 to perform as an H.323 VoIP gateway using RAS, perform the following tasks:
Numbering-type matching is used in dial-peer configuration mode to match on a number type for a dial peer call leg.
To configure numbering-type matching using the numbering-type command in dial-peer configuration mode, enter the following commands starting in global configuration mode:
| Step | Command | Purpose |
|---|---|---|
| 1. | ||
| 2. | Enters the dial peer configuration mode to configure a VoIP or POTS peer. |
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| 3. |
When a called number is received and matched to a POTS dial peer, the matched digits are stripped and the remaining digits are forwarded to the voice interface. A new command called the digit strip option makes this behavior an option. Digit stripping is enabled by default.
To disable digit strip for a dial peer, enter the following commands in global configuration mode:
A dial peer defines the characteristics associated with a call leg. Dial peers are used to apply attributes to call legs and to identify call origin and destination. Attributes applied to a call leg include QoS, codec, VAD, and fax rate. A call leg is a discrete segment of a call connection that lies between two points in the connection. All of the call legs for a particular connection have the same connection ID.
There are two different types of dial peers:
A POTS dial peer points to a voice-port on the router, and the destination of a VoIP dial peer points to the destination IP address of the voice-router that terminates the call.
Complete the following procedures to configure call legs using the translation-rule command:
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TimeSaver You should configure your translation rules before you apply rules to your dial peer call legs. |
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Note Applying translation rules to more than one dial peer call leg in your end-to-end call is not recommended. |
Step 2 To apply a rule to an inbound POTS call leg, enter the following commands in global configuration mode:
Step 3 To apply a rule to an outbound VoIP call leg, enter the following commands in global configuration mode:
Step 4 To apply a rule to a VoIP call that originates from an H.323 node, enter the following global command:
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Note There can only be one global voip-incoming translation-rule. |
Step 5 To apply a translation rule to an outbound POTS call leg, enter the following commands in global configuration mode:
| Step | Command | Purpose |
|---|---|---|
| 1. | Enters the dial-peer configuration mode to configure a POTS dial peer. |
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| 2. | ||
| 3. | Specifies the translation tag for inbound called or calling number. |
Step 6 Enter the test translation-rule command.
This section contains the following configuration examples:
The following example shows how to configure a Cisco 3600 series router as an H.323 gateway:
For RAS to work on an H.323 gateway, you need to configure a corresponding RAS gatekeeper. The following example shows a Cisco 3600 series router configured as a RAS gatekeeper. For more information about configuring gatekeepers, refer to the Cisco IOS Release 11.3(6)NA2 document, Configuring the Cisco AS5300 for Voice Service Provider Features.
| Command | Purpose |
|---|---|
Displays information about the rules that have been configured for a specific translation name. |
This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.0 command reference publications.
In Cisco IOS Release 12.0(1)T or later, you can search and filter the output for show and more commands. This functionality is useful when you need to sort through large amounts of output, or if you want to exclude output that you do not need to see.
To use this functionality, enter a show or more command followed by the "pipe" character (|), one of the keywords begin, include, or exclude, and an expression that you want to search or filter on:
command | {begin | include | exclude} regular-expression
Following is an example of the show atm vc command in which you want the command output to begin with the first line where the expression "PeakRate" appears:
For more information on the search and filter functionality, refer to the Cisco IOS Release 12.0(1)T feature module titled CLI String Search.
To match on a number type for a dial peer call leg, use the numbering-type command. To remove the specified number-type use the no form of this command.
Syntax Description
Command Modes
Command History
Examples
The following example shows to configure the POTS dial peer for network usage:
Related Commands
To display the contents of all the rules that have been configured for a specific translation name, use the show translation-rule global configuration command.
Syntax Description
The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. Range is 1 to 2147483647. |
Command Modes
Command History
Usage Guidelines
This command gives detailed information about the configured rules under this rule name. If the name tag is not entered, a complete display of all the configured rules will be shown.
Examples
The following example shows output for the show translation-rule command:
Related Commands
To test the execution of the translation rules on a specific name tag, use the test translation-rule global configuration command. To disable, use the no form of this command.
Syntax Description
Defaults
No default behavior or values.
Command Modes
Command History
Examples
The following example shows output for the test translation-rule command:
Related Commands
To apply a translation rule to a calling party number or a called party number for both incoming and outgoing calls, use the translation-rule global configuration command. To disable, use the no form of this command.
translation-rule name-tag precedence input-searched-pattern substituted-pattern [match-type | substituted-type]
Defaults
No default behavior or values.
Command Modes
Command History
Usage Guidelines
When configuring your dial peers, you are provided with an option called the translation rule. This rule applies a translation rule to a calling party number [Automatic Number Identification (ANI)] or a called party number [Dial Number Information Service (DNIS)] for both incoming and outgoing calls within Cisco H.323 voice-enabled gateways. Also, the rule allows translation of the type of number.
Examples
The following example applies a translation-rule. If a called number starts with 5272205 or 72205, then the translation-rule 21 will use the rule command to forward the number to 14085272205 instead.
In the next example, if a called number is either 14085272205 or 014085272205, then after the execution of the translation-rule 345, the forwarding digits will be 72205. If the match-type is configured and the type is not "unknown," then the dial peer matching will be required to match input string numbering type.
Related Commands
For calls that originate from H.323-compatible clients, use the voip-incoming translation-rule global configuration command. To disable, use the no form of this command.
voip-incoming translation-rule name-tag {calling-number | called-number}
no voip-incoming translation-rule name-tag
Syntax Description
Defaults
No default behavior or values.
Command Modes
Command History
Usage Guidelines
With this command, all IP-based calls will be captured and handled, depending on either the calling or called number to the specified tag-name.
Examples
The following example identifies the rule set for calls that originate from H.323-compatible clients:
Related Commands
There are no new or modified debug commands for this feature.
AAA—Authentication, Authorization, and Accounting. AAA is a suite of network security services which provides the primary framework through which access control can be set up on your Cisco router or access server.
ANI—Automatic number identification.
CAS—Channel associated signaling.
CCAPI—Call Control Application Programming Interface
CLI—Command Language Interpreter. The basic Cisco IOS configuration and management interface.
dial peer—An addressable call endpoint. In Voice over IP (VoIP), there are two types of dial peers: POTS and VoIP.
DNS—Domain name system used to address translation to convert H.323 IDs, URLs, or e-mail IDs to IP addresses. DNS is also used to assist in the location of remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains.
endpoint—An H.323 terminal or gateway. An endpoint can call and be called. It generates and/or terminates the information stream.
gatekeeper—A gatekeeper maintains a registry of devices in the multimedia network. The devices register with the gatekeeper at startup, and request admission to a call from the gatekeeper.
The gatekeeper is an H.323 entity on the LAN that provides address translation and controls access to the LAN for H.323 terminals and gateways. The gatekeeper may provide other services to the H.323 terminals and gateways, such as bandwidth management and locating gateways.
gateway—A gateway allows H.323 terminals to communicate with non-H.323 terminals by converting protocols. A gateway is the point at which a circuit-switched call is encoded and repackaged into IP packets.
A H.323 gateway is an endpoint on the LAN that provides real-time, two-way communications between H.323 terminals on the LAN and other ITU-T terminals in the WAN, or to another H.323 gateway.
H.323—An International Telecommunication Union (ITU-T) standard that describes packet-based video, audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the conferencing system, and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual protocol.
H.323 RAS—Registration, admission, and status. The RAS signaling function performs registration, admissions, bandwidth changes, status and disengage procedures between the VoIP gateway and the gatekeeper.
IVR—Integrated voice response. When someone dials in, it responds with a prompt to get a personal identification number (PIN), and so on.
multicast—A process of transmitting PDUs from one source to many destinations. The actual mechanism (that is, IP multicast, multi-unicast, and so forth) for this process might be different for LAN technologies.
multipoint-unicast—A process of transferring PDUs (Protocol Data Units) where an endpoint sends more than one copy of a media stream to different endpoints. This might be necessary in networks which do not support multicast.
node—A H.323 entity that uses RAS to communicate with the gatekeeper. For example, an endpoint such as a terminal, proxy, or gateway.
POTS—Plain old telephone service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the PSTN.
PSTN—Public switched telephone network. PSTN refers to the local telephone company.
QoS—Quality of service, which refers to the measure of service quality provided to the user.
RAS—Registration, admission, and status protocol. This is the protocol that is used between endpoints and the gatekeeper to perform management functions.
TDM—Time-division multiplexing. Technique in which information from multiple channels can be allocated bandwidth on a single wire based on preassigned time slots. Bandwidth is allocated to each channel regardless of whether the station has data to transmit.
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.
VTSP—Voice telephony service provider.
zone—A collection of all terminals (tx), gateways (GW), and Multipoint Control Units (MCU) managed by a single gatekeeper (GK). A zone includes at least one terminal, and can include gateways or multipoint control units (MCUs). A zone has only one gatekeeper. A zone may be independent of LAN topology and can be comprised of multiple LAN segments which are connected using routes or other devices.
Posted: Fri Jan 17 00:20:20 PST 2003
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