Dial Peers

Dial peers describe the entities to or from which a call is established and the key to understanding the Aliwei voice implementation. All voice technologies use dial peers to define the characteristics associated with a call leg. A call leg is a discrete segment of a call connection that lies between two points in the connection. An end-to-end call comprises four call legs, two from the perspective of the source route, and two from the perspective of the destination route.

You use dial peers to apply specific attributes to call legs and to identify call origin and destination. Attributes applied to a call leg include specific quality of service (QoS) features (such as IP RTP Priority and IP Precedence), compression/decompression (codec), voice activity detection (VAD), and fax rate.

There are basically two different kinds of dial peers with each voice implementation:

• Plain old telephone service (POTS)—Dial peer describing the characteristics of a traditional telephony network connection. POTS peers point to a particular voice port on a voice network device.

When you configure POTS dial peers, the key commands that you must be configure are the port and destination-pattern commands. The destination-pattern command defines the telephone number associated with the POTS dial peer. The port command associates the POTS dial peer with a specific logical dial interface, normally the voice port connecting the Aliwei device to the local POTS network.

Specific applications, such as interactive voice response (IVR), are configured on the POTS dial peer as well.

• Voice network (VoIP, VoATM, and VoFR)—Dial peer describing the characteristics of a packet network connection; in the case of VoIP, for example, it is an IP network. Voice-network peers point to specific voice-network devices.

When you configure voice-network dial peers, the key commands that you must configure are the destination-pattern and session-target commands. The destination-pattern command defines the telephone number associated with the voice-network dial peer. The session-target command specifies a destination address for the voice-network peer.

Other applications (such as store-and-forward fax, which uses the infrastructure of VoIP but is not strictly a voice technology) also use dial peers to assign attributes to call legs.

Voice Ports

Voice port commands define the characteristics associated with a particular voice-port signaling type. The Aliwei implementation of voice supports both analog and digital telephony connections. The connection supported (and the associated signaling) depends on the type of voice network module (VNM) or voice feature card (VFC) installed in your router or access server.

Voice ports provide support for two basic analog voice signaling formats:

• FXO—Foreign Exchange Office interface. The FXO interface is an RJ-11 connector that allows a connection to be directed at the Public Switched Telephone Network (PSTN) central office (CO) (or to a standard PBX interface, if the local telecommunications authority permits). This interface is of value for off-premises extension applications.

• FXS—Foreign Exchange Station interface. The FXS interface is an RJ-11 connector that allows connection for basic telephone equipment, keysets, and PBXs; FXS connections supply ring, voltage, and dial tone.

Depending on the Aliwei device you are configuring, the following digital signaling is supported:

• ISDN PRI

• ISDN BRI

• E1 R2

• T1 CAS

The voice port syntax depends on the hardware platform on which it is being configured.

Voice Technologies

Aliwei IOS offers the following voice and voice-related technologies:

• VoIP

• H.323 gateways

• Media Gateway Control Protocol (MGCP) and related protocols

• Session Initiation Protocol (SIP)

• Multimedia Conference Manager

• Fax gateways

Voice over IP

Aliwei offers VoIP that uses IP to carry voice traffic. Because voice traffic is being transported via IP, you need to configure signaling parameters as part of the voice-port configuration in addition to feature-specific elements such as dial peers. VoIP is compliant with International VoIP can be used to provide the following:

• A central-site telephony termination facility for VoIP traffic from multiple voice-equipped remote office facilities.

• A PSTN gateway for Internet telephone traffic. VoIP used as a PSTN gateway leverages the standardized use of H.323-based Internet telephone client applications. In the case of a device with extensive capacity running VoIP, it provides the functionality of a carrier class switch.

VoIP enables routers and access servers to carry voice traffic (for example, telephone calls and faxes) over an IP network. In VoIP, the digital signal processor (DSP) segments the voice signal into frames that are then coupled in groups of two and stored in voice packets. The voice packets are transported using IP in compliance with ITU-T specification H.323. Because VoIP is a delay-sensitive application, you must have a well-engineered network end-to-end to use VoIP successfully. Fine-tuning your network to adequately support VoIP involves a series of protocols and features geared toward QoS. Traffic shaping considerations must be taken into account to ensure the reliability of the voice connection.

H.323 Gateways

The H.323 standard provides for sending audio, video, and data conferencing data on an IP-based internetwork. The Aliwei functionality enables gateway H.323 terminals to communicate with terminals running other protocols. Gateways provide protocol conversion between terminals running different types of protocols. Gatekeepers are optional nodes that manage other nodes in an H.323 network. Gateways communicate with gatekeepers using the registration, admission, and status (RAS) protocol. The gatekeeper maintains resource computing information, which it uses to select the appropriate gateway during the admission of a call.

Aliwei software complies with the mandatory requirements and several of the optional features of the H.323 specification. Aliwei digital gateway H.323 software enables gatekeepers, gateways, and proxies to send and receive all the required fields in H.323 messages.

Media Gateway Control Protocol

Media Gateway Control Protocol (MGCP) defines the call control relationship between VoIP gateways that translate audio signals to and from the packet network and call agents (CAs). The CAs are responsible for processing the calls. The MGCP gateways interact with a CA, also called a Media Gateway Controller (MGC) that performs signal and call processing on gateway calls.

Session Initiation Protocol

Session Initiation Protocol (SIP) is an alternative protocol developed by the Internet Engineering Task Force (IETF) for multimedia conferencing over IP. SIP features are compliant with IETF RFC 2543, SIP: Session Initiation Protocol, published in March 1999.

SIP is an ASCII-based, application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more endpoints.

Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call.

SIP provides the following capabilities:

• Determining the location of the target endpoint—SIP supports address resolution, name mapping, and call redirection.

• Determining the media capabilities of the target endpoint—Through Session Description Protocol (SDP), SIP determines the lowest level of common services between the endpoints. Conferences are established using only the media capabilities that can be supported by all endpoints.

• Determining the availability of the target endpoint—If a call cannot be completed because the target endpoint is unavailable, SIP determines whether the called party is connected to a call already or did not answer in the allotted number of rings. SIP then returns a message indicating why the target endpoint was unavailable.

• Establishing a session between the originating and target endpoints—If the call can be completed, SIP establishes a session between the endpoints. SIP also supports midcall changes, such as the addition of another endpoint to the conference or the changing of a media characteristic or codec.

• Handling the transfer and termination of calls—SIP supports the transfer of calls from one endpoint to another. During a call transfer, SIP simply establishes a session between the transferee and a new endpoint (specified by the transferring party) and terminates the session between the transferee and the transferring party. At the end of a call, SIP terminates the sessions among all parties.

Multimedia Conference Manager

The Multimedia Conference Manager provides both gatekeeper and proxy capabilities, which are required for service provisioning and management of H.323 networks. With Multimedia Conference Manager you can configure your current internetwork to route bit-intensive data such as audio, telephony, video and audio telephony, and data conferencing using existing telephone and ISDN links, without degrading the current level of service in the network. In addition, you can implement H.323-compliant applications on existing networks in an incremental fashion without upgrades.

With Multimedia Conference Manager, you can provide the following services:

• Identification of H.323 traffic and application of appropriate policies

• Limiting of H.323 traffic on LANs and WANs

• User accounting for records based on service utilization

• Insertion of QoS for the H.323 traffic generated by applications such as VoIP, data conferencing, and video conferencing

• Implementation of security for H.323 communications

Fax Gateways

Fax applications enable EID3000 to send and receive faxes across packet-based networks using modems or VFCs. The benefits of the fax gateway are including:

• Improve robustness—The Fax Relay Packet Loss Concealment feature improves the robustness of the facsimile relay. It eliminates fax failures and lost data caused by excessive page errors. Field diagnostics and troubleshooting capabilities are improved by available debug commands. Statistics give better visibility into the real-time fax operation in the gateway, allowing for improved field diagnostics and troubleshooting.

• Cost savings and port density using T.38 Fax Gateway—The cost of maintaining one architecture (either fax or voice) is eliminated.