Accelerating the Deployment of Voice Over IP (VoIP) and Voice over ATM - SEMINAR

Definition and Overview

 

          The economic advantages of packet voice are driving both the access and core voice networks away from circuit switching towards packet. The industry continues to debate whether the future of these packet networks will be based on pure ATM, pure Internet protocol (IP), IP over asynchronous transfer mode (ATM) or a combination thereof. There are advantages to both ATM and IP and reasons for choosing each. This tutorial will explore the role of next-generation switches which, as they become widely adopted for both access and core networking, must be able to handle voice traffic over both IP and ATM networks for future extensibility as the debate continues and must have the features necessary to interwork with existing public switched telephone network (PSTN).

1. Introduction

 

          Carriers are moving voice services to packet networks both to reduce upfront and operational costs and to provide more value-added services in an increasingly competitive environment. A recent study by a major carrier found that packet equipment was 70 percent less expensive than traditional voice equipment, and data access lines were 60 percent to 80 percent cheaper than voice lines. Maintenance of packet networks was 50 percent less expensive, while provisioning was 72 percent lower. However, consolidation of voice from the PSTN onto packet networks has, in the past, proven difficult and therefore has happened very slowly.

          While it is clear that Voice over packet is growing, there is still considerable debate about whether the underlying network technology will be ATM or IP. At the edge of the network the choice, driven primarily by the regional Bell operating companies (RBOCs), is ATM. An ATM–dominated access network is clearly in the works because until recently IP did not provide the quality of service (QoS) guarantees that are so important for voice. Although QoS protocols such as Difference Service and resource reservation protocol (RSVP) have been implemented, most of today's IP traffic is actually being carried over ATM. However, in the long term with the recent success of MPLS it appears that pure IP over lambda may be the winner. And certainly, IP at the application layer and the desktop is a more than just a viable near-term situation.  

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2. Voice over Packet Architecture

 

          Carrier-class next-generation switches need to be high-capacity fault-tolerant TDM and Voice over Packet switches. They must be designed to significantly enhance the economics of providing traditional TDM–based voice and data services as well as help service providers migrate to a packet-based telecom network (based on VOIP and VOATM) and generate new competitive services. Service providers deploying next-generation switches can cap their investment in traditional circuit switches and migrate to a converged switching infrastructure that allows them to reduce the number of overlay network platforms and provide profitable voice and data services over packet networks. See Figure 2.1.

 

          Since most of the core packet networks today are ATM–based, but most likely migrating to IP–based, the most future-proof investment is in next generation switches that can be deployed to transport voice on both ATM and IP networks supporting protocol layers as outlined in Figure 2.2.

 

3. voice over internet protocol

 

         

          Support for voice communications using Voice over Internet Protocol is a protocol optimized for the transmission of voice through the internet or other packet switching networks. Voice over internet protocol is often used abstractly to refer to the actual transmission of voice (rather than the protocol implementing it). VoIP is also known as IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband.

 

          Voice over IP protocols carry telephony signals as digital audio typically reduced in data rate using speech compression techniques and encapsulated in a data packet stream over IP. The two major competing standards for Voip are the ITU standard H.323 and the IETF standard Sip.  In voice networks where everything is under the control of the network operator or (Telco), H.323 is the choice of protocol.h.323 references a large number of protocols for speech encoding, call setup, signaling and data transport etc.

                       

VOIP work:

          When you speak at the handset or a mike or a microphone, your voice generates electrical signals inside the gadget. These are analog signals i.e. the voltage level can take up any value within a range.

          The analog signal is converted to a digital signal using an algorithm implemented by the device you are using. It can be a stand-alone VoIP phone or a softphone running on your PC. If you are using an analog phone, you will need a Telephony Adapter (TA) for this purpose. The digitized voice is arranged in packets (i.e. collection of bytes) and sent over the IP network.

          The data is channeled through gateways and servers to the destination. If the called number is on the PSTN, the server opens a connection to the PSTN and routes your call there.

          While going to the PSTN or at the end device of a VoIP connection, the voice is again brought back to its analog form so that it is perceptible to a human ear.

          During the entire process a protocol like SIP or H.323 is used to control the call (e.g. setting up connection, dialing, disconnecting etc.) and RTP is used for reliable transmission of data packets and maintain Quality of Service.

Benefits of using voice over internet protocol:

          voice over internet protocol has become especially attractive to consumers given the low-cost, flat-rate pricing of the public Internet.

          VoIP is the ability to make telephone calls and access service over IP–based data networks with a suitable QoS and superior cost/benefit to PSTN–based calls. Today, most of the VoIP implementations are carried over ATM–based transport as shown in the second column of Figure 2.

          The benefits of implementing VoIP are mostly consumer-based and can be divided into the following three categories:

·    Cost reduction—IP is everywhere. It is on our desktops and it is what the Internet is based on. Many people view the Internet as a "free transport" for data and voice services. With the introduction of Net2Phone and other similar "free" services, many people are now making phone calls over the Internet. In addition, businesses and individuals have turned to higher-quality commercial products and services to make voice calls based on IP. The prevalence of IP nodes and the abundant supply of better IP–based switches and routers continue to reduce the cost of providing VoIP.

·    Simplification and consolidation—An integrated infrastructure that supports all forms of communication could allow more standardization and could reduce the total equipment complement. The differences between the traffic patterns of voice and data offer further opportunities for significant efficiency improvements. Universal use of IP for all applications, voice and data, holds out the promise of both reduced complexity and more flexibility.

·    Advanced applications—Even though basic telephony and facsimile are the initial applications for VoIP, the longer-term benefits are expected to be derived from multimedia and multiservice applications. For example, Internet commerce solutions can combine World Wide Web access to information with a voice call button that allows immediate access to a call center agent from a PC. In addition, voice is an integral part of conferencing systems that could also include shared screens, white boards, etc. Combining voice and data features into new applications will provide the greatest returns over the longer term.

          Utilizing an IP–based network for voice traffic can offer advantages to consumers of reduced costs, simplification, and consolidation due to the proliferation of IP–based applications and devices at the desktop. These advantages are compelling for consumers and are driving service providers to consider voice over internet protocol implementations. In contrast, Voice over packet network is the ATM–based  offers distinct advantages directly to service providers and are still much more prevalent today.

4. Voice over asynchronous transfer mode (VOATM)

 

          ATM, from the start, was designed to be a multimedia, multiservice technology. Although ATM has been accepted by service providers for its ability to deliver high-speed data services, until recently its potential for deploying voice services was overlooked. With the competitiveness of today's market though, network operators and service providers have been continuously striving to reduce operating costs and lift network efficiency and have turned to the ATM network to achieve these goals.

 

          With hundreds of millions of dollars of ATM equipment infrastructure in the United States alone, service providers have recognized that significant economies of scale can be achieved if the data traffic and voice traffic are integrated onto a single network. In order to achieve this, service providers have started to use the circuit emulation services (CES) of ATM switches to carry full or fractional E-1/T-1 circuits between end points. These CES mechanisms treat voice as a constant stream of traffic encoded as a constant bit rate (CBR) stream. In actuality though, voice is a combination of bursts of speech and silence and this increases the complexity of Voice over packet

          Asynchronous transfer mode  results in a connection oriented network for communication .Asynchronous transfer mode  has its own 3 dimensional model consisting of three layers physical layer deals with the physical medium voltages, bit timing and other various issues.  Asynchronous transfer mode layer deals with the cells and cell transport .It defines the layout of cell and what header field mean. Asynchronous transfer mode adaptation layer interface segments these input packets and transmits  the cells individually and reassembles at the other end .

          The ATM Forum and International Telecommunications Union (ITU) came up with several advanced mechanisms to improve the efficiencies of transporting voice traffic, including:

·      ATM trunking using AAL–1 for narrowband services

·    ATM trunking using AAL–2 for narrowband services

·    IP over ATM (AAL–5)

·    Loop emulation service using AAL–2

5. Design Considerations for Voice over Packet

 

          Adding voice to packet networks requires an understanding of how to deal with system level challenges such as

1.     Interoperability

2.     Call control and signaling

3.     Voice encoding

4.     Delay

5.     Echo

6.     Reliability

7.     Density and

8.     Performance of all the elements that make up the next-generation switching platform.

6. Elements of a Next-Generation Switching Platform

 

 

          The vision for a next-generation switching platform is a distributed architecture in which media gateway/bearer transport platform, signaling, call control, and application elements are divided into separate logical network components, communicating with one another through the use of intraswitch protocols such as Megaco, media gateway control protocol (MGCP), and SCTP/M3UA. This distributed model allows service providers to scale their network to support hundreds of thousands of subscriber ports per node. In this concept, voice traffic is directed between the traditional voice network and the new packet-based networks by the media gateway. The call control is handled by a softswitch, and the features and services are handled by an application platform. In reality, the softswitch (or call control platform) may support some of the more popular services without requiring a separate application platform. An example of this type of service is 7/10 digit routing, which would be handled directly by the call control platform. Other examples of where the application platform may not be involved are caller name delivery, local number portability (LNP), and E-800 service. These services are already implemented in the PSTN using service control points (SCPs). In these cases, the call control platform will send intelligent network (IN)/transactional capabilities application part (TCAP) queries over the signaling system 7 (SS7) network to existing SCPs.

 

          Some vendors enable one or more of these logical network elements to be deployed on the same physical platform. There are some inherent advantages to this "integrated" model especially with platforms that support up to 100,000 subscriber ports (DS–0s) per bearer platform/media gateway, and allow efficient execution of the softswitch and signaling gateway software. Benefits also include cost savings and deployment and operation simplicity. In the "integrated" model, the need for intraswitch protocols such as Megaco and MGCP are not required; however interswitch protocols such as RTP/UDP/IP (for MG to MG) and BICC (for SG to SG) are always required for interoperability with the other ends. See Figure 4 for relevant inter-switch protocols.

 

 7. switching plotform/media gateway

           

            Sometimes referred to as a media gateway, the switching/bearer transport platform is hardware that sits at the edge of a network and takes in a packet and/or circuit containing voice or data traffic and switches it to a voice or data network. Media gateways come in many different flavors depending on the breadth of definition. The most popular consist of Class 4 and Class 5 replacement functionality and a voice over digital subscriber line (VoDSL) gateway. Media gateways are part of the physical transport layer and are controlled by a call control engine or softswitch (also called a media gateway controller), which provides instructions to direct voice traffic. Media gateways are at the heart of the transformation of the voice network, as they are essential to migrating voice traffic onto a packetized network. As part of packetizing voice traffic, a media gateway adapts (by using compression and echo cancellation) the packetized traffic, creates and attaches an IP header and/or ATM header, and sends the packet through the network according to instructions provided by the softswitch.

          While a media gateway can be physically located almost anywhere within the network, depending on the network architecture and the features it is intended to support, all media gateways share certain features including the following:

·    Scalability—A media gateway needs to be able to scale to support hundreds of thousands of telephone calls (called DS–0s, running at 64 Kbps per line) to parallel the scalability of the existing PSTN switches.

·    Support for several types of access networks—Needed support includes wireless, fiber, cable, and copper. In addition to electrical interfaces, a media gateway needs to support a variety of optical interfaces (including OC–3, OC–12, OC–48, and OC–192 speeds).

·    Carrier-class reliability—Also known as five nines (99.999 percent) reliability (i.e., less than five minutes of downtime per year) and network equipment building standards (NEBS) certification (the Telcordia quality rating for meeting environmental stress tests), reliability is extremely important to service providers because it enables them to fulfill customer contracts. Most carriers cite reliability as the impetus to transform their current architecture.

·    Interworking functionality—Media gateways are capable of supporting multiple voice and data interface protocols and compatibility between them by converting circuit traffic to packet traffic and vice versa.

·    Interoperability—Most networks are a compilation of multivendor solutions, making interoperability essential for success.

·    Control support—To enable communication between the media gateway and a softswitch. The most common languages (or protocols) emerging for communication between these devices are MGCP and Megaco.

·    Switching—A media gateway must handle switching and media processing, based on an ATM, IP, or TDM switching fabric.

·    Voice transportation—There are 3 transport standards used for transporting voice traffic: TDM (traditional circuit-switch method), ATM AAL–1/AAL–2, and IP–based RTP/RTCP (over ATM or pure–IP transport).

Technical challenges in transferring voice over packet architecture:

          A packetized approach to transmitting voice faces a number of technical challenges that spring from the real-time or interactive nature of the voice traffic. Some of the challenges that need to be addressed include the following

·        Echo—Echo is a phenomenon where a transmitted voice signal gets reflected back due to unavoidable impedance mismatch and four-wire/two-wire conversion between the telephone handset and the communication network. Echo can, depending on the severity, disrupt the normal flow of conversation and its severity depends on the round-trip time delay if a round-trip time delay is more than 30 ms the echo becomes significant making normal conversation difficult.

·   End-to-end delay—Voice traffic is most sensitive to delay and is mildly sensitive to variations in delay (jitter). It is critical that end-to-end delay is minimized to hold interactive communications. Delay can interfere with the dynamics of voice communication, in the absence of noticeable echo, whereas in the presence of noticeable echo, increasing delay makes echo effects worse. When delay reaches above 30 ms, echo canceller circuits are required to control the echo.

·   Packetization delay (or cell construction delay)—Packetization delay is the time taken to fill in a complete packet/cell before it is transmitted. Normal G.711 pulse code modulation (PCM) encoded voice samples arrive at the rate of 64 Kbps, which means it can take approximately 6 ms to fill the entire 48-byte payload of an ATM cell. The problem can be addressed either with partially filled cells or by multiplexing several voice calls into a single ATM virtual circuit channel (VCC).

·   Buffering delay—Sometimes, due to delay in transit, some cells might arrive late. If this happens the ATM segmentation and reassembling (SAR) function provided by the adaptation layer might have to under run with no voice data to process which results in gaps in conversation. To prevent this, the receiving SAR function would accumulate a buffer of information before starting the reconstruction. In order to ensure that no under runs occur the buffer size should exceed the maximum predicted delay. The size of the buffer translates into delay, as each cell must progress through the buffer on arrival at the emulated circuit's line rate. This implies that the cell delay variation (CDV) has to be controlled within the ATM network.

·   Silence suppression—Voice, by its nature, is variable. In fact, a typical conversation has a speech activity factor of about 42 percent due to pauses between sentences and words where there is no speech in either direction. Also, voice communication is half-duplex, which means that one person is silent while the other speaks. One can take advantage of these two characteristics to save bandwidth by halting the transmission of cells during these silent periods. This is known as silence suppression.

·   Compression algorithms—G.726 adaptive differential pulse code modulation (ADPCM) and G.729 adaptive code excited linear prediction (ACELP) are the two major compression algorithms that are used. The benefit of compression is efficient use of bandwidth. Most voice packets are transmitted today using G.711 encoding that does no compression and therefore adds further delay

8. Signaling Gateway

 

          A signaling gateway is hardware and software that provides the connection from a softswitch and media gateway to the SS7 network. The signaling gateway receives/sends the call control instructions needed between the SS7 network and the softswitch, typically through stream control transmission protocol (SCTP) and MTP Level-3 user adaptation layer (M3UA) protocols. This allows the softswitch to process and communicate call control instructions to a media gateway. A signaling gateway can either stand-alone or be integrated with a softswitch/media gateway. In the traditional circuit-switched telephone network, a legacy switch provides the interface directly to the SS7 world, essentially acting as a signaling gateway

9. The Softswitch/Media Gateway Controller

 

          A softswitch, also referred to as a "call agent" or "media gateway controller," is software that provides the call control and signaling for the next-generation network. The softswitch moves the service intelligence out of the switch into a database or application server, connects those databases, and ultimately provides the "brains" or operating system for the next-generation voice network. A softswitch ensures that a call is routed through the network to the proper destination and that features from the existing advanced intelligent network (AIN) such as 1-800 and LNP, as well as new multimedia services, are applied to calls as appropriate. While the softswitch architecture is similar to the AIN databases in an SCP, a softswitch provides more robust functionality and is distinguished by providing control to more than one type of switch—including TDM, ATM, IP, etc.—while today's AIN controls only TDM–based switches. This architecture is inherently more flexible and scalable than the architecture of today's circuit switches.

There is significant debate in the industry about the definition of a softswitch, its role within the network, how it should interface with other gateways and softswitches, and how it should interface with the IP and SS7 networks. At the most basic level, a softswitch must contain call-control features and a signaling interface to the SS7 network. Call control relates to the setup and teardown of calls, including service selection ("which services apply to this call?") and call routing ("where will this call be sent?"). In addition, a softswitch must provide call authentication ("what calls is this line allowed to make?"), authorization, and accounting services by accessing information available in the existing SS7 network. The SS7 signaling interface, which allows the softswitch to communicate with today's SS7 network, is in some cases distributed to a stand-alone hardware system called a signaling gateway. Today's softswitches typically operate on the Sun Solaris operating system and include features such as the following:

Media independence— to make the software agnostic regarding the switching fabric (ATM, IP, TDM, etc.)

Interoperability—with multiple vendors' media gateway products, the existing PSTN, and off-the-shelf hardware platforms

Reliability—to carrier standards (five 9s of reliability)

Support for multiple signaling and control protocols—including emerging and established standards such as ISUP, BICC, SIP, and MEGACO/H.248

Scalability—to meet carrier network requirements, supporting thousands of call attempts, also known as busy hour call attempts (BHCA) and simultaneous calls

Open application programming interfaces (APIs)—or "hooks" into third-party software applications and services

10. Application Server (AS) and Services

 

 

          Finally, without services, next-generation switches would not be able to generate the voice revenue that currently provides 80 percent of overall service provider revenue. The following Class-4 and Class-5 services need to be supported by these switches. As stated before, several of these services may be implemented in the softswitch (call control platform) without the need for an external AS. The more complex services such three-way conferencing may require the need for an AS with multimedia support. Until ASs become more capable in terms of supporting more complex services and providing robust easy-to-use service creation environments (SCEs), the need to deploy these services outside of the softswitch environment is less compelling.

·        Dial tone

·        Basic dialing

·        Basic 7/10 routing

·        Announcements

·        Billing record creation

·        Call blocking/allow

·        Call transfer

·        Call forward

·        CF busy

·        CF no answer

·        3-way calling

·        Calling name delivery

·        Calling number delivery

·        Premium rate: 900/976

·        Selective call reject

·        Selective call accept

·        Remote call forwarding

·        Speed dialing 30

·        Anonymous call reject

·        Caller ID block

·        Automatic callback

·        Automatic recall

·        Call waiting

·        Calling identity delivery on call waiting

·        Customer-originated trace

·        Distinctive ringing/call waiting

·        Selective call acceptance

·        Selective call forwarding

·        Selective call rejection

11. Conclusion

 

          Voice packet telephony is a reality today, although, as an industry, there still is a lot of work ahead. The larger incumbent carriers are starting the migration to packet telephony on the trunk side first (Class-4 tandem) and will eventually migrate to the access side (Class 5). This migration model is similar to the migration from analog switches to digital switches, which started in the late 1970s, and offers a proven path for migration to new technologies. The full migration to packet-based Class 5 systems will happen when the inner network becomes packet-based and when differentiated Class-5 services become available. And the services must go beyond currently available PSTN–based services for packet telephony to become truly compelling. On this journey, the debates over VoIP and VoATM will continue. While VoATM makes sense today for some carriers, especially the larger incumbents, VoIP is the longer-term goal especially with MPLS–based QoS becoming available. For some carriers, VoIP is the answer today as the consumer benefits are persuasive.Service providers looking to deploy VoP will be best served if they choose a solution that addresses the issues of interoperability, call control and signaling, voice encoding, delay, echo, reliability, density, and performance of all the elements that make up the switching platform. And they should look for solutions that deal with these issues for TDM switching, as well as VOIP and VOATM.

 

references

 

 

Book: 

·        “Computer Networks” by Andrew S. Tanenbum.

 

Websites:

·        www.google.com

·        www.iec.org

·        www.ietf.com

·        www.ieee.com