Voice over Frame Relay, IP and ATM


The Case for Cooperative Networking

by Gil Biran,
Vice President of Research and Development,
RAD Data Communications Ltd.

Table of Contents

  • Abstract
  • The Trend toward the Integrated Voice and Data Network
  • The Nature of the Data Network and its Implications for Voice
  • Voice over Frame Relay
  • Voice over IP
  • Voice over ATM
  • Interoperability Standards with Room for Interpretation
  • Need for Interworking
  • Conclusion
  • Abstract

    There’s a lot of talk about the suitability of Frame Relay, IP and ATM for carrying voice and the advantages of one over the other. It is clear that there are advocates of all three technologies and that each is suited to particular needs and environments. That being the case, they will all probably be around for some time to come. It is therefore becoming apparent that there will be a need for interworking between them.

    In this paper we will discuss the voice-enabling mechanisms employed by all three technologies and solutions for interoperability.

    The Trend toward the Integrated Voice and Data Network

    In the last few years, data networks have been growing at a much faster rate than voice networks, mainly due to the growth of the Internet. Soon the amount of data traffic will exceed that of voice traffic. As a result of this trend, more and more voice is being sent over data networks (Voice over Frame Relay, Voice over IP and Voice over ATM) than data is being sent over voice networks (via V.34 and V.90 modems).

    When Frame Relay was introduced in the early 1990s, the data technology was not originally designed to carry voice. Despite valid reservations about the reliability of voice over frames, the promise of “free voice” eventually proved too alluring. Soon users were experimenting with transporting voice over their Frame Relay devices while equipment vendors worked overtime to make the promise of quality voice over Frame Relay (VoFR) a reality.

    As the public Internet exploded in the mid-1990s and users began implementing IP-based networks, the call for voice over IP (VoIP) grew louder. Here, too, equipment manufacturers are developing products to enable inexpensive, universal voice over data networks.

    Carriers, however, were caught in a dilemma. Could they afford to cannibalize their highly profitable public switched telephone network? Could they not afford to capitalize on the demand for digital voice? The drama is just unfolding.

    Although significant progress has been made in engineering packet networks (Frame Relay, IP and ATM) to carry voice as well as data, today’s market is demanding a true convergence of these technologies into a single and ubiquitous communications service without being limited by the underlying technology. The next challenge, then, is to develop interconnection and interworking standards in order to deliver voice services ubiquitously over Frame Relay, IP and ATM.

    The Nature of the Data Network and its Implications for Voice

    The Nature of the Data Network
    Frame Relay, IP and ATM are known as packet or cell switching technologies. This is in contrast to the public telephone network, which is a circuit switching technology, designed to carry voice transmissions. Frame Relay and IP insert data into variable-sized frames or packets. ATM chops data into small cells, which facilitates fast switching of data through the network.

    The packet switching and cell switching networks perform statistical multiplexing. That is, they dynamically allocate bandwidth to various links based on their transmission activity. Since bandwidth is not reserved for any specific path, the available bandwidth is allotted according to network needs at any particular time.

    Compare this to the traditional voice (or circuit switching) network, in which a path is dedicated to the transmission for the duration of the call, which is sent in a continuous bit stream. The line is monopolized by a call until it is terminated, even when the caller is put on hold and during periods of silence. Although this guarantees reliable and immediate transmission of voice, it results in very inefficient use of bandwidth. A line that is dedicated to the telephone cannot be utilized by other data even when there are no voice transmissions.

    Originally designed to handle bursty data traffic, packet switching networks (except for ATM) are inherently less efficient than the circuit switching network in dealing with voice. To achieve good voice quality, the delay of voice packets across the network must be minimal and fixed. Due to the shared nature of the packet/cell switching network, it might take time for transmissions to travel across the network. A transmission can be delayed because of network congestion. For example, it might “get stuck” behind a long data transmission that delays other packets. Network congestion can also result in dropped packets, which also detrimentally affects the integrity of voice transmissions.

    Voice-Enabling the Data Network
    Unlike most data applications, voice is very sensitive to delay. Good voice quality provides a faithful recreation of the conversation, with the same tone, inflection, pauses and intonation used by the speakers. Long and variable delays between packets result in unnatural speech and interfere with the conversation. Dropped packets result in clipped speech and poor voice quality. Fax transmissions are even more sensitive to the quality of the transmission and are less tolerant of dropped packets than voice.

    One way to deal with the problem of delay and congestion is to add bandwidth to the network at critical junctures. Although this is feasible in the backbone, it is a costly and ineffective solution in the access arena, defeating the “bandwidth sharing” benefits of packet networks. The best solution is to implement mechanisms at the customer premises, access node and backbone which manage congestion and delay – without increasing bandwidth – such as setting priorities for different types of traffic. Therefore, smart access equipment was developed, that could implement procedures to reduce network congestion and the delay of voice packets without adding bandwidth.

    Voice over Frame Relay (VoFR)

    Of the three popular packet/cell technologies (Frame Relay, IP and ATM), Frame Relay is the most widely deployed. Frame Relay is commonly used in corporate data networks due to its flexible bandwidth, widespread accessibility, support of a diverse traffic mix and technological maturity.

    Frame Relay service is based on Permanent Virtual Connections (PVCs). Frame Relay is appropriate for closed user groups and is also recommended for star topologies and when performance needs to be predictable. VoFR is a logical progression for corporations already running data over Frame Relay.

    Voice Frame Relay access devices (VFRADs), such as RAD’s MAXcess integrated bandwidth manager, integrate voice into the data network by connecting the router (or using the integrated router available on certain MAXcess models), SNA controller and the PBX at each site in the corporate network to the Frame Relay network.

    Many VFRADs, such as RAD’s MAXcess, employ sophisticated techniques to overcome the limitations of transporting voice over the Frame Relay network without the need to add costly bandwidth.

    Note: These techniques are discussed below for Frame Relay. Implementation for IP and ATM is discussed in the Voice over IP (VoIP) and Voice over ATM (VoATM) sections, respectively.

    The VFRADs’ prioritization schemes “tag” different applications according to their sensitivity to delay, assigning higher priority to voice and other time-sensitive data such as SNA. The VFRADs let the higher priority voice packets go first, keeping the data packets waiting. This has no negative effect on data traffic, as voice transmissions are relatively short and, being compressed, require very little bandwidth. They can therefore slip into the data network alongside the heavy graphics, payroll information, e-mail and other data traffic without perceptibly encumbering overall network performance.

    Frame Relay service providers have also begun to offer different Quality of Service (QoS). Users can purchase the highest quality of service, Real-Time Variable Frame Rate, for voice and SNA traffic. The second quality Frame Relay service, Non-Real Time Variable Frame, is typically purchased for LAN-to-LAN and business class Internet and intranet traffic. The lowest quality of service, Available/Unspecified Frame Rate, is used for e-mail, file transfer and residential Internet traffic. In addition, the VFRAD can be configured to assign less sensitive traffic with a Discard Eligibility (DE) bit. These frames will be dropped first in case of network congestion.

    The MAXcess and other VFRADs incorporate fragmentation schemes to improve performance. Data packets are divided into small fragments, allowing higher priority voice packets to receive the right-of-way without waiting for the end of long data transmissions. The remaining data packets in the data stream are momentarily halted until the voice transmission gets through.

    The down-side of fragmentation is that it increases the number of data frames, thereby increasing the number of flags and headers. This increases overhead and reduces bandwidth efficiency. RAD’s FR+ application provides an enhanced fragmentation mechanism which fragments data frames only in cases where voice packets arrive at the switch during a data transmission. Otherwise, the long data frames are sent intact.

    Controlling Variable Delay
    Variation in the arrival times between packets, also called jitter, causes unnatural-sounding voice instead of a smooth voice stream. If a packet does not arrive in time to fit into the voice stream, the previous packet is replayed. This can seriously detract from voice quality. To avoid the effect of jitter, VFRADs such as the MAXcess detain each packet in a jitter buffer, giving subsequent packets time to arrive and still fit into a natural voice flow. Since the jitter buffer adds to the overall delay of voice transmissions, the optimal jitter buffer should fit the network’s differential delay. Better access devices, like RAD’s MAXcess, employ adaptive jitter buffering, which continuously monitors the network delay and adjusts the queuing period accordingly.

    Voice Compression
    Voice compression allows the packet switching network to most effectively carry a combination of voice and data sessions without compromising voice quality. Since Frame Relay access is usually at data rates of 56/64 kbps, low bit-rate voice compression algorithms such as ITU G.723.1 and G.729A permit the greatest number of simultaneous multiple calls while maintaining high quality voice. Vendors such as RAD, which have implemented voice compression algorithms in their Frame Relay access devices, can offer greater bandwidth savings, reduced network congestion, and high quality voice transmissions.

    Silence Suppression
    In a telephone conversation, only about 50% of the full duplex connection is used at any given time. This is because, generally, only one person talks while the other person listens. In addition, voice packets are not sent during interword pauses and natural pauses in the conversation, reducing the required bandwidth by another 10%. Silence suppression frees this 60% of bandwidth on the full duplex link for other voice or data transmissions.

    Echo Cancellation
    Echo cancellation improves the quality of voice transmissions. It eliminates the echo that results from the reflection of the telephony signal back to the caller, which can occur in a 4-wire to 2-wire hybrid connection between the VFRAD and the telephones or PBX. The longer it takes the signals to return to the caller, the more perceptible the echo.

    Voice over IP (VoIP)

    Internet Protocol (IP) is a connectionless protocol in which packets can take different paths between the endpoints and all paths are shared by packets from different transmissions. This enables efficient allocation of network resources, as packets are routed on the paths with the least congestion. Header information makes sure that the packets reach their intended destinations and helps reconstruct the messages at the receiving end. To ensure QoS, however, all packets should use the same path. IP headers are large (20 bytes) as compared to the headers of Frame Relay frames (2 bytes) and of ATM cells (5 bytes). Headers therefore add a lot of overhead to IP traffic.

    IP networks employ the same types of bandwidth-saving schemes as the Frame Relay network, including fragmentation, jitter buffering, prioritization, voice compression, silence suppression and echo canceling.

    Prioritization techniques used for VoIP are different from those employed by Frame Relay access devices. Prioritization is directly related to QoS. The key IP QoS protocol was RSVP, which allowed the sender to request a certain set of traffic-handling characteristics for traffic flow, but was not widely adopted. Today, the Intserv working group of the IETF is developing a simpler, more promising solution. The Differentiated Services Model uses the Type of Service (ToS) octet field of the IP header to classify traffic at the borders between the customer and service provider or Internet service providers (ISPs). Currently, there is still no viable QoS for IP services.

    IP fragmentation is performed in a similar fashion as Frame Relay fragmentation. Fragmentation, although required to reduce the overall delay of voice traffic, adds a lot of overhead to IP transmissions due to the large size of IP headers. IP voice traffic therefore consumes 50% more WAN bandwidth than Frame Relay voice traffic. However, as IP matures, header compression and improved routers will eliminate these shortcomings.

    Voice compression is vital in Voice over IP because traffic usually travels over low-speed links. Some small and medium-sized enterprises, for example, may be connected to the virtual private network (VPN) at only 28.8 kbps. Microsoft Netmeeting, for example, a popular voice application for PCs and laptops, supports ITU G.723.1 voice compression for transmission over dial-up modems. The ITU G.723.1 standard for voice compression over IP ensures toll quality voice.

    Jitter buffer, silence suppression and echo cancellation techniques are similar to those employed in VoFR. Echo cancellation is extremely important in VoIP, which often suffers from long network delays.

    Voice over ATM (VoATM)

    Asynchronous Transfer Mode, or ATM, is a multiservice, high speed, scalable technology. It is a dominant switching fabric in carrier backbones, supporting services with different transfer characteristics. ATM simultaneously transports voice, data, graphics and video at very high speeds. On the down side, ATM services are expensive and not yet universally available.

    Large enterprises are increasingly connecting headquarters and main offices to the wide area network via broadband links such as ATM to accommodate their vast amounts of voice and data transmissions, such as heavy graphics, payroll information and voice and video conferencing.

    Fragmentation is built into ATM, with its small, fixed-size, 53-byte cells. Very fast ATM switches speed data through the ATM network. The high bandwidth associated with ATM reduces congestion problems, providing extremely reliable service. Carriers can therefore promise customers Quality of Service (QoS), stipulated in Service Level Agreements (SLAs).

    ATM prioritization is implemented through QoS parameters. ATM was designed from the outset to carry voice as well as all types of data. ATM Adaptation Layer 1 (AAL1) protocol in ATM’s Constant Bit Rate (CBR) service was the de facto standard for VoATM. However, this protocol proved inefficient for voice applications. CBR, the highest quality class of ATM service, provides Circuit Emulation Service (CES), which transmits a continuous bit stream of information. This allocates a constant amount of bandwidth to a connection for the duration of a transmission. Although it guarantees high quality voice, CES monopolizes bandwidth that could be used for other applications. In addition, in the interest of reducing delay, CES might send the fixed-size ATM cells half empty rather than waiting 6 milliseconds for 47 bytes of voice to fill the cell. This wastes over 20 bytes of bandwidth per ATM cell. Dynamic Bandwidth Circuit Emulation Service (DBCE S) is a variation of CES. DBCES does not send a c onstant bit stream of cells, but transmits only when there is an active voice call (off hook). However, like in CES, the cells might remain partially empty. Therefore, using AAL1 for VoATM increases the overhead of voice transmissions and wastes bandwidth.

    AAL2’s Variable Bit Rate (VBR-RT) service, as specified in ITU-T recommendation I.363.2, emerged as the standard of choice for VoATM. The structure of AAL2 allows for the packing of short packets (1 to 45/64 bytes), also called minicells, into one or more ATM cells. (This resembles Frame Relay’s and IP’s variable sized fragments.) In contrast to AAL1, which has a fixed payload, AAL2 enables a variable payload within cells. This functionality provides a dramatic improvement in bandwidth efficiency over structured or unstructured circuit emulation using AAL1. In addition, AAL2 supports voice compression and silence suppression and allows multiple voice channels with varying bandwidth on a single ATM connection.

    Voice compression is not necessary in pure-ATM networks, which enjoy ample bandwidth. However, in hybrid ATM-Frame Relay networks (for example, with ATM headquarters and Frame Relay branches), voice compression is required since Frame Relay uses voice compression. ATM must therefore be equipped to support voice compression that will work with VoFR equipment at the remote site.

    Interoperability Standards with Room for Interpretation

    The interworked voice and data network of the future promises the best of all network worlds: the installed base of Frame Relay, the speed and quality of ATM and the ubiquity of IP. Currently, fragmentation techniques in Frame Relay, IP and ATM are quite similar, but prioritization techniques, signaling protocols and voice compression algorithms are not compatible. Progress is being made toward standardization within each protocol and interworking between them, but considerable work remains.

    The Frame Relay Forum has defined standards for sending voice over Frame Relay. Implementation Agreement FRF.11 presents basic Frame Relay definitions but allows a lot of leeway regarding voice switching and negotiation between different VFRADs. This necessitates a proprietary solution, precluding interoperability between equipment of different vendors.

    Phase 2 of FRF.11, which deals with interoperability issues, is not yet implemented. As long as Frame Relay is based on Permanent Virtual Connections (PVCs), which establish fixed links, there is little incentive for vendors to interoperate. With PVCs, all nodes are known, and same-vendor equipment is used on all end devices. If Switched Virtual Connections (SVCs) were more widely used paths would be defined dynamically, and interoperability would be elevated to a market requirement. In the meantime, a proprietary interworking method is an acceptable solution in private corporate networks.

    ITU H.323 defines interoperability standards for voice and multimedia applications over IP. It determines endpoint negotiation and information format. It does not, however, address encoding, prioritization or security. And like other standards, some of the H.323 definitions are subject to interpretation. Therefore, H.323 does not guarantee interoperability between equipment of different vendors. Some IP vendors are collaborating on an interoperability profile based on ITU H.323 and on the upcoming H.225.0 Annex G standard. Its goal is to achieve interoperability between gateways and gatekeepers of different vendors, enabling deployment of different IP platforms at either end of the network. Once the profile is released, however, extensive testing will be required to determine the actual level of inter-vendor compatibility.

    Variable Bit Rate AAL2 is recommended for VoATM. Although it sounds promising, the AAL2 standard is not yet completely developed. It will most likely suffer from the same inter-vendor interoperability problems as the VoFR and VoIP solutions, at least in the short run.

    The Need for Interworking

    Since a comprehensive standard has not been adopted for any one technology, it is unrealistic to expect the emergence of interoperability standards between technologies in the near future. Interworking solutions will therefore have to be proprietary. It is essential that the interoperability be transparent to the users, who want to communicate through the network efficiently and without concern for the technological issues involved.

    Due to the lack of interoperability standards for voice communications over Frame Relay, IP and ATM, vendors must develop proprietary interworking solutions.

    There are many situations in which interworking between technologies is required within a corporate network. For example, corporations that are running data and voice over Frame Relay might require VoIP to extend the network to remote locations that don’t have a Frame Relay infrastructure without deploying additional equipment. This may also be required for telecommuters working from home, salespeople working from hotel rooms and resellers that want to access information. Consider the configuration below, in which IP is extended to hotels or residential locations.

    voice interworking

    Interworking between Frame Relay and IP is vital in this scenario. This is easier said than done, however, as Frame Relay and IP use different standard voice compression algorithms (ITU G.729 and ITU G.723A, respectively). There is also a discrepancy between signaling methods. The FRF.4 recommendation for switched virtual circuits is not widely used as the basis for Frame Relay voice switching. In fact, there is no standard for voice switching in Frame Relay networks. In any case, it does not interwork with VoIP voice switching based on the H.323 protocol stack.

    In large corporate networks with many remote branches using Frame Relay service and VoFR, there is a need for high speed ATM service in order to support the required amount of traffic at company headquarters. In this case there is a need for interworking between Frame Relay and ATM in general and VoFR and VoATM in particular.

    Another reason for the desirability of multiple voice technology support over the same platform is to enable migration to different technologies without losing the initial investment in existing equipment.

    RAD is developing a pre-standards strategy to facilitate interworking between Frame Relay and IP. The strategy will provide a migration path from Frame Relay to IP technologies, which can be an important advantage when IP services become available. RAD is currently developing an interworking solution between Frame Relay and IP. The VoFR-VoIP product will perform signaling conversion and negotiate with the remote IP product in order to choose a common voice compression algorithm and other parameters.


    Without a doubt, the data revolution will only gain momentum in the coming years, with more and more voice traffic moving onto data networks. Vendors of voice equipment will continue to develop integrated voice and data devices based on packetized technology. Users with ubiquitous voice and data service integrated over one universal infrastructure will benefit from true, seamless, transparent interworking between voice and all types of data. RAD Data Communications will remain in the forefront of industry efforts to provide universal services.