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May 2001

Scot Robertson

How A VoP Call Is Made


Voice over Packet (VoP), telecommunications using high-density media gateways, is attracting enormous attention and investment. As standard phone service migrates from the continuous-circuit public switched telephone network (PSTN), to a packet-switched VoP technology, explaining both the need for this conversion and its complexities from both a technical and a financial perspective presents challenges to service providers.

From a user's standpoint, placing a VoP call that travels over a private network or even the Internet is not all that different from placing a traditional call that travels over the PSTN. In fact, that lack of perceived difference is critical to the success in ultimately converting to a packetized approach. Customers expect seamless quality of service (QoS) to accompany successful VoP deployment. That means whenever VoP technology replaces the PSTN and does not offer a new service or convenience, such as the mobility of Cellular phones, the quality of VoP system service must closely approximate that of the PSTN such that service quality and reliability are transparent to the end user. Only by delivering the appropriate level of quality will manufacturers and service providers accomplish their goal of deploying voice access over a packetized network.

And that won't be easy. Because even though call transmission may be transparent to the user, from an equipment and routing standpoint, VoP call placement, setup, and transmission is quite different and complex. Routing a call over a network, or over the Internet, requires the use of extensive network equipment and software resources; and those resources are being deployed incrementally, in part because QoS is paramount. As a result, service providers are installing hybrid systems that utilize both approaches.

In the near term, the "vision" of VoP systems will be supported by these hybrid deployments that use the PSTN to provide worldwide access to the end user -- while packetized, VoP infrastructure is developed for long-haul services and competitive local exchange carrier (CLEC) access to end users with the assumption that these hybrid systems will eventually convert to full VoP deployment. Unfortunately, the implementation of total VoP systems is impeded by today's infrastructure limitations and by the economics that impact service providers. For example, although the accessibility of the Internet is growing rapidly, the absence of substantive QoS controls and standards deter user adoption of voice over Internet protocol (VoIP) services. Because the voice quality of VoIP is inconsistent and often unreliable, VoIP systems are today deployed primarily in limited, private networks, or campus and enterprise-based local area network (LAN) systems such as the IP PBX. These systems provide the advantage of a single LAN connection for both voice and data services, but voice calls almost always connect back to a circuit-switched PSTN for off-campus connections.

Deregulation, compounded with the proliferation of service suppliers, adds to the mix because as suppliers announce new opportunities and services, it is clear that there is no "one size fits all" VoP solution. To make it even more complicated, whole new technologies that converge voice and data are being implemented across the communication infrastructure. For example, voice and data packets are transported through trunking gateways, or high-capacity multi-channel access concentrators. They provide an essential single point of access by converting voice signals from the PSTN to packetized data and vice-versa. As long as a conversion is required gateways are necessary regardless of the subscriber's connection type such as the public phone system, a mobile phone, VoIP, VoCable, VoDSL, PBX, or other system. Trunking gateways are designed to process thousands of calls at a time, connecting users, campuses/workgroups, and customer premises sites.

It is clear that a total conversion to a VoP infrastructure will ultimately prevail because of the imperatives of the inherent economic advantages. However, some of the factors that contribute to inconsistent and incremental deployment must be overcome. These include variations in network service capabilities, access and capacity limitations, and limited voice quality due to disruptive network characteristics such as latency and packet loss. Finally, interoperability between manufacturers, and the need for consistent standards for product development and deployment, must continue to be addressed. That cooperation has accelerated as service providers and operators perceive the opportunity for new revenues. Nevertheless, when infrastructure, common access, and performance issues are resolved, industry experts believe rapid and widespread deployment and adoption of integrated networks based on common standards will prevail.

A VoP call is defined as any call in which any part of the communication is carried over a packet network. Although many PSTN or circuit-switched calls are digitized and converted from analog to digital, the signal is sent as a continuous stream of bits at a rate of 64 kilobits per second. For a packetized network call the key criterion requires fixed or variable length packets of bits sent over the network, not the continuous stream.

So, what happens when someone makes a VoP call? The process of sending voice over a network (or sending voice packets over a network) is compatible with using an Internet service provider (ISP) to transmit voice packets. The packet flow begins after the call is placed. There are many variations, but one scenario starts with a VoIP call through a local-area network (LAN) or IP PBX originating at the end-users premises. The call is routed via a local packet network, often DSL or frame relay, as a series of packets to the local exchange carrier's central office. The central office is the user's access point to the PSTN. At this point the voice packets may be converted via a trunking gateway into continuous pulse code modulation (PCM) bit stream for connection into the PSTN.

The call then travels through the digital trunk lines to the Internet service provider's (ISP) point of presence (POP). Digital trunk lines are the physical (wire) and logical connections between circuit switches across which telephone network travels in PCM form. The trunk lines terminate in the trunking gateway, or Internet gateway processor, that connects to an ISP's packet network. The trunking gateway compresses the voice data and creates network packets that are transmitted into the packet network or Internet.

Once on the packet network, the call is routed to the end destination premise gateway through a virtual circuit. A virtual circuit is a communications link that appears to the user as a dedicated point-to-point circuit. In reality, the packetized data can be sent over different physical paths through a network to its destination. The virtual circuit determines where the packetized speech for the specific phone call should be routed and enables the receiving gateway to verify receipt, enable packet re-order, identify, and correct discrepancies. In fact, virtual circuits illustrate the key benefits of packet-based networks, and they are cheaper and faster. In the receiving gateway, packets are converted back to a continuous stream of data and connection with the local PSTN for transmission to the receiver.

There are variations to this theme depending upon the nature of both the sending and receiving networks. In fact, even part of a virtual circuit may require interim conversion to a circuit-switched system where the packets are converted to a continuous stream and subsequently re-packetized. Again, the system employs gateways to enable the conversions.

An interesting variation is with calls through the use of wireless networks. Most wireless networks use digitized systems similar to speech packets in a VoP call. However, the wireless networks use a variety of protocols to compress speech, one of the essential steps in the packet creation process. As a result, the call first has to route through a transcoder and rate adaption unit (TRAU) that converts speech from the user of a wireless system into the digital code required by the IP-based network compatible with standards such as H.323. This adds both additional complexity of equipment and can increase latency, because speech compression decoding and encoding incurs additional latency.

Latency is a problem inherent in VoP systems and is primarily caused by network delay or by the functions that occur within the gateway. However, any additional latency due to transcoding is a problem. Consequently efforts are underway to enable networks to enable a common IP-based protocol across wireless and wireline systems.

Finally, the packetization process in which real-time speech and real-time control protocols convert, sequence, and then transmit packets to the network must also enable the signal to be reconstructed before it reaches the receiving user. These packets constantly enter and exit the processing components, and arrive delayed, distorted, and out of sequence, far different from PSTN continuous bit-stream systems. Hardware and software at the users end gateway detects and processes the voice signals, compensates for distortion and provides the call routing. The end result is these hybrid elements combine to work with the Internet protocol (or other network-based protocol) and the PSTN to deliver the phone call transparently to the user. 

Scott Robertson is product line manager for Remote Access Products at Analog Devices. Analog Devices is a semiconductor company that develops, manufactures, and markets high-performance integrated circuits (ICs) used in signal-processing applications. 

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