Managing voice and data services

Integrated voice and data is a new service that is offered by service providers. The architecture design to deploy managed voice and data services is based on many factors. One of these factors is the required customer premises equipment (CPE), which is based on the type of business customer.

Two

general types of business customers exist. One is a business customer with fewer than 100 users, such as a doctor’s office, an insurance agent’s office, or a small home office.

These businesses are normally single-site locations that require telephony services, Internet access, firewall, and Virtual Private Network (VPN) services. Typically, these businesses do not have older networking protocols, such as AppleTalk or IPX, and they do not have a full- time support staff to maintain their own private network. A service provider can support these services with an IAD on the customer premises, such as a Cisco 2400.

The second type of business customer is an enterprise customer. An enterprise customer has a large installed base of devices that supports many flavors of protocols, sophisticated routing designs, multiple T1s, and back-hauling needs. An enterprise customer needs a multiservice platform, such as the Cisco 2600 and 3600.Many of these enterprise businesses have their own large private networks and their own full-time staff to maintain their multiservice network. However, because of various reasons, such as fast growth and economics, many of these large customers are outsourcing some or all of their services to service providers.

Integrated Access Architectures

Traditionally, service providers offer TDM services that connect a customer’s PBX to an IXC Class 4 switch, which provides long distance voice services. Many of these service providers are currently switching from using a TDM-based infrastructure to using a packet-based infrastructure, either IP or ATM. This approach allows for a more efficient method to provide voice transport and also helps to integrate voice and data services over one access link to the customer premises.

Managed Voice and Data Services Using AAL2

AAL2, referred to as VoAAL2 in a voice network, can integrate the voice and data services offered to the customer. Alternatively, a service provider can begin with an IP-based infrastructure and build out a VoIP call agent architecture to support voice and data services to their customers, which is a more common approach today. Both technologies, VoIP and VoAAL2, offer the value of integrating voice and data while achieving efficient bandwidth use.

This section provides an AAL2 architecture that can provide trunking and integrated access services. By using AAL2, many capabilities can be obtained within the service provider’s ATM network:

Dynamically change from voice to fax demodulation

AAL2 Type 3 cells for reliable dual tone multifrequency (DTMF) relay

Dynamically change the compression rate to G.711 for fax calls in mid call

Indicate end of speech burst for background noise generation during silence periods at the egress ATM switch

Transport up to 248 voice calls with different compression schemes within one or more ATM permanent virtual circuit (PVC)

This architecture provides a Class 4 interconnect replacement, which enables an enterprise to bypass the local Tandem Switch.

Fundamentals of AAL2

The AAL2 protocol has two layers:

Service specific convergence sublayer (SSCS)

Common part sublayer (CPS)

The SSCS encodes different information streams for the transport by AAL2 over a single ATM connection. The information streams might be active voice encodings, silence insertion descriptors, dialed digits, or fax. SSCS can provide error control on critical information (CAS signaling and dialed digits)by using a 10-bit CRC. This is called an AAL2 Type 3 cell. The SSCS segments the information that is being passed from a higher layer application, such as samples of voice from a digital phone into a number of units of data, and submits these units of data to the CPS for transmission. The length of the segmented data can be between one and the maximum length supported by the CPS connection, which is either 45 or 64 bytes. At the SSCS receiver, the units of data are reassembled back into the information before being passed to the higher layer application.

The second layer, the CPS, is specifically responsible for transporting end-to-end connections across the network.

Although AAL2 with its three-byte packet header introduces some inefficiency for small packets, the improvement that is reached by having no padding more than offsets this minor inefficiency.

Start field are described here:

Start field enables efficient packing of the voice packets over a single ATM virtual circuit. The Offset field is a six-bit pointer within the Start field that points to the position of the first CPS packet that follows the OSF.A sequence number protects the order of the Offset field. If a Start field parity error exists, all the CPS packets that are associated with the Start field are discarded.

CID (Channel ID) Identifies the end user, which is referred to as the SSCS entity in the International Telecommunication Union (ITU)AAL2 specifications. The CID allocates the value 1 to exchange layer management peer-to-peer procedures, such as set-up negotiations. CID enables the multiplexing of up to 248 user channels, whereas some CID values are reserved for other uses, such as peer-to-peer layer management.

For example, if 8 E1s terminated on an MGX, 240 CID values would be used.

LI (Length Indicator) Identifies the length of the CPS packet. The default payload length is 45 bytes, and an optional maximum length of 64 bytes can be selected. The maximum length is channel specific.

UUI (User-to-User Indication) Provides two functions: It conveys specific information transparently between two end points (e. g., CPS or SSCS entity)and distinguishes between the different users, such as SSCS entities and layer management users.

HEC (Header Error Control) Discards the rest of the CPS packets until the next Start field. As a result, not all voice users residing on the single ATM virtual channel are affected by other end-user errors, which results in a higher end-to-end efficiency.

The CID is an important concept in AAL2.CIDs provide a binding between an endpoint and an AAL2 connection. This is the mechanism that binds the TDM traffic to the ATM traffic.

For example, if a service provider needs to provision 100 DS0s between two sites for one of its enterprise customers,100 CIDs are created across the ATM network. Furthermore, a unique coder-decoder (codec) type is assigned to each individual DS0 because the codec type is assigned to each CID through an AAL2.For example, individual customers in a multitenant building can each support multiple compression schemes over a single T1 access link.

Each CID is configured and includes the following parameters:codec type, profile type, voice activity detection (VAD),DTFM Tones, and packet period for G.729.For example, to transmit DTMF tones transparently across the ATM PVC,DTMF must be enabled in the CID.

An AAL2 profile is a mechanism that the MGX 8850 uses to assign the compression and encoding scheme of the AAL2 trunking service. A profile is defined by a profile type, which is either an ITU standard or a custom type and a number. These profiles need to match on both ends of the network for the two end devices, such as PBXs, to interoperate. A profile is configured for each CID. For example, if the profile type is ITU and the profile number is 1, you must use G.711.In other words, the profile type and the profile number identify the compression type.

CID enables the use of sub cell multiplexing, which provides many of the benefits of AAL2.

If you use G.711,sub cell multiplexing does not provide any value because G.711 already uses an 80-byte packet. The real advantage of sub cell multiplexing is the G.729 encoding scheme. If you use G.729 with a packetization period of 30 milliseconds, three 10-byte packets of payload from one DS0 are packed into one ATM cell. Therefore, the efficiency of packing the voice sample into the ATM cell is increased threefold, and instead of 34 bytes of padding, only 14 bytes exist in the ATM cell. Assuming that VAD provides an additional 50 per cent of bandwidth savings, G.729 sub cell multiplexing uses approximately 6 kbps of bandwidth per DS0 channel of voice traffic. This is a significant amount of bandwidth savings.

 

This chapter provided an overview of managed voice and data services using ATM technology. AAL2 is an important component in providing a managed service using ATM. Because of efficient bandwidth use and the ability to transport different traffic types, AAL2 is used in trunking applications, such as interconnecting mobile wireless sites. Today, many service providers use a pure IP-based architecture to support this service.