1 The Proliferation of VOIP by Estevan Macias, Global Technology Resources, Inc., Boulder, Colorado Introduction Voice over IP is finally living up to the promise of inexpensive, ubiquitous, feature-rich, voice service. This has resulted in reduced cost and increased productivity. Several factors have come together to make this possible. Some of these factors include the increased bandwidth to the end-user, certain regulatory legislation, technical innovation and price-driven competition. This rapid proliferation and adoption is not without its issues. Given standard Cosler- Decoder s (CODECs), Quality of Service (QOS) is required yet no service guarantee is offered over the Internet. This has led to newer, innovative, low-bit-rate CODECs both open source and proprietary. This has also let to a stratification of VOIP types; commercial and personal use. Softswitches are a means of transporting toll-quality VOIP on a large scale. These have supplanted traditional PBXs (Private Business Exchange) such as DMS (Digital Multiplex system) 100s and Dyfinity in many newer networks. Scales of economies are achieved when minimal signaling channels are used to transport a maximum number of Barrier channels. Today s commercial VOIP continues to use medium-complexity, standards-based CODECs. Where GoogleTalk and Skype make use of proprietary custom developed low-bit-rate CODECs. Another issue of note is the less-flexible Telcos slow to adopt softswitch technology complicating interoperability. Given the substantial investment made by many telcos in older network architectures, and in an effort to maximize return on investment, are retrofitting their existing network to accommodate VOIP. Clearly, newer, more flexible Telcos with modern networks have a competitive advantage. Some work-arounds include converting VOIP to analog, transport it across a legacy telco networks then re-convert the analog to VOIP. This is the opposite approach first used when VOIP was the new and unaccepted technology. Several newer competitive local exchange carriers, such as Vonage, Sunrocket and Packet8 use VOIP nearly exclusively. E911 service, is challenging in a technically advanced, potentially mobile environment. The issue is fundamentally one of mobility. Voice over IP over (a set of communication standards for wireless local area networks WLAN) is further complicating the location issue. Dual-mode phones (cellular and voip over ) offer even greater flexibility and cost savings but further compound the issue. The solution to this problem is currently being crafted by innovative companies such as Intrado. The following examines these issues and some potential solutions revolving around the deployment and use of VOIP as a personal and business communications tool.
2 Quality Of Service Quality of Service ( QOS ) is an umbrella of data mechanisms used to ensure the prioritization of one type of data traffic over another. This is, in essence, an attempt to make a multi-access system perform in a deterministic manner, to create order out of chaos. In order to accomplish this, a tag or some value must be assigned to specific IP packets and/or underlying Ethernet frames, while others are left unassigned or assigned a different value. This notion is referred to as marking. Another issue is each intervening network device must ensure that IP traffic is adhering or conforming to the markings. This concept is referred to a policing. In order for QOS to have an effect, QOS must be end-to-end. Given the complexity of today s networks and the Internet, this requires a Herculean effort. Further complicating the issue is that there are several QOS paradigms in use in today s networks. In its worst case, QOS can mis-prioritize traffic and have an adverse impact on the intended traffic. Class of Service (COS) Class of service is an OSI (Open Systems Interconnection a standard protocol for communications systems design) layer-2 construct, that is, it operates at the network switch level, and as such, is found on Ethernet frames. It is comprised of 3 binary bits for a total of eight values (0 7). Routine traffic is ascribed the value of 0, the lowest value, while Flash and Priority, value 6 and 7 respectively, are reserved. VOIP barer traffic is typically assigned the COS value of 5 and VOIP call control is typically assigned the COS value of 3. These COS values are conveyed from Ethernet device to Ethernet device, until the frames encounter a router, which operates at OSI layer-3. The router removes the target IP packet from the Ethernet frame, takes note of its intended destination and of its QOS markings then re-encapsulates the packet into the appropriate WAN format, including the QOS, it then sends the packet to the next upstream router. IP Type of Service (TOS) The initial attempt at OSI layer-3 packet marking was Type of Service. TOS is very much a layer-3 equivalent of COS. It is comprised of 3 bits and therefore has a total of 8 possible values. The lowest value is 0 while the highest value is 7. This has proven to be a significant limitation to the scalability of QOS, particularly in Telco and large-scale networks. This led to a revamping of IP packet standards and gave rise to DSCP. TOS, like the TCP/IP protocol suite has been in existence in the late 1960s. Version 4 of TCP/IP has undergone several retrofits to include security (IPSEC) and Network Address Translation (NAT) and RFC1918 to address globally unique IP address depletion.
3 IP version 6, designed from the ground up to address these concepts is another topic altogether. VOIP, however, in the not-so-distant future, will need to be transported over IP v6. Differentiated Service Code Point (DSCP) Integrated Services was the standards bodies first attempt to update TCP/IP to include QOS. This effort gave rise to the Resource Reservation Protocol (RSVP) and common open policy server (COPS) protocols. These have not come into common usage as compared to constructs taken form Differentiated Services; the most notable being DSCP. DSCP has largely supplanted TOS as the preferred layer-3 QOS method. DSCP is comprised of 6 binary bits within the IP packet, for a total of 64 values AF class 1 AF class 2 AF class 3 AF class 4 Low Drop Medium Drop High Drop Fragmentation and Interleaving Toll quality VOIP barer traffic is typically sent in Real Time Protocol (RTP) packets atop User Datagram Protocol (UDP). This collection is sent to its destination further encapsulated within an IP packet. These VOIP packets are typically small and uniform in size. As such, these are not a significant burden to the network, unto themselves. The most significant attribute required by VOIP of a network is consistent response time. Consistent response time is achieved, partially, by ensuring that small VOIP packets are not kept waiting behind large packets of another traffic type. Typically, the transmission of data packets is made most efficient by transmitting packets of the largest size supported by the technology being used. For example, the Maximum Transmissible Unit (MTU) of Ethernet is 1518 bytes, where the MTU of Frame Relay is Fragmentation is the process of segmenting the packets into smaller transmissible units. Interleaving is the process of transmitting VOIP packets in between the segmented packets of another traffic type. These techniques allow for more uniform response time and subsequently, more uniform voice quality. Other QOS constructs include technology specific QOS marking and policing methods such as Asynchronous Transfer Method (ATM) traffic classifications which include Unspecified Bit Rate (UBR), Variable Bit Rate (VBR), Real-Time VBR(rt-VBR) and Circuit Emulation Service (CES). Another technology specific QOS method is Frame
4 Relay Traffic Shaping (FRTS) for Frame Relay. These methods are slowly being migrated away from toward MPLS which is described in a subsequent section. Mean Opinion Score (MOS) VOIP sound quality is quantified in terms of Mean Opinion Score (MOS). MOS is measured in a scale from 0 to 4.5, with 4.5 being the theoretical maximum sound quality achievable by VOIP. Toll quality voice is typically accepted as 3.8 or higher. As the name implies, MOS was, at least initially, derived from the opinions of a cross-section of test subjects. Modern VOIP analysis tools can simulate VOIP traffic, analyze several metrics, and determine with great accuracy, the MOS afforded by a specific data network or network path. Some of these VOIP analysis tools include software products made by Ixia, Chariot, Agilent, Cisco Systems and Network General The above graphic displays a most estimate graph generated by Ixia. Of note is the consistent, high quality result generated by the network against which this test was taken. This was generated across a multi-gigabit LAN. CODEC Coder-Decoder is typically abbreviated as CODEC. Various CODECs provide different MOS. These are inherent in the CODEC and are determined under ideal bandwidth conditions. These are typically software constructs though may be etched in silicon in the form of application specific integrated circuits (ASIC). There are standards-based CODECs and custom-developed CODECs both open source and proprietary. In rare cases, two different VOIP system manufactures may implement standards-based codes and not have the same performance or in extreme cases interoperability issues. The CODEC used greatly determines the sound quality of phone calls in a given environment. A given environment, from a network perspective, includes factors such as bandwidth, types of other traffic and quantity of other traffic types.
5 The manufacture of a given VOIP system must balance the sound quality of a selected CODEC with the computational complexity of that CODEC. It serves no real purpose to create the perfect sounding low-bit-rate CODEC if it requires a Cray IV supercomputer. The VOIP system manufacture must also take into careful consideration, the amount of bandwidth with which the CODEC was intended to function. Standards bodies have ratified several CODECs in common use. These include G711, G723, G726, G728, G729, G729a, and G729b. These have varying inherent sound qualities, computational complexity and optimal bandwidths. Of note, regarding Conjugate-Structure algebraic Code Excited Linear Predicative (CS-CELP) CODECs is that these codes were modeled after spoken language and tend to attenuate other sounds. This has the unfortunate side-effect of not working well with certain languages which make use of pops and clicks. Further Music-on-hold and sounds such as car crashes and gunshots tend also to be attenuated. As such, these types of CODECs may be less desirable for use where phone calls may be recorded and used as admissible evidence. Nearly all VOIP manufactures support multiple CODECs for use under differing network conditions. Calls made using the G711 CODEC are generally considered to consume approximately 80 kbps per call. Although this may seem modest, this can fill up a T-1 rapidly especially when adding in traditional data traffic. Co-mingling Voice and Data on a single transport is one of the greatest benefits of VOIP. Calls made using the G729 CODEC are generally considered to consume approximately 20 kbps per call. Proprietary codes are typically designed solve a specific problem not addressed by standards-based CODECs or add features not included in standards-based CODECs. Open sources CODECs are typically associated with open-source software-based implementations of PBXs such as Asterisk. Implementers of VOIP systems are afforded a myriad of choices of VOIP systems. The best performance is achieved across a private network, with ample bandwidth, QOS and use of a wideband CODEC (G711). The best cost savings is achieved with a proprietary, low-bit-rate CODEC running across the open Internet by using a free Internet connection discovered by war driving. Clearly, striking a balance between cost and performance is how most VOIP implementations are deployed. Testing tools may be use to validate the performance of design parameters. The above-mentioned points demonstrate the need for a detailed design phase as part of any VOIP deployment. Telco Service Providers Many of today s service providers are scrambling to remain competitive and profitable in light of intense competition and blindingly fast technical developments. One of the business challenges faced by Telcos is maintaining balance between differentiating themselves from their competition while maintaining interoperability with other telcos. They are faced with the schizophrenia of wanting to be technically superior to their competition yet interoperable with their competition.
6 The Telecom Act of 96 and the DOJ (Department of Justice) Modified Final Judgment from 1982 have served to increase competition, at least in some aspects. The greatest value retained by the ILECs (Incumbent Local Exchange Carrier) is the installed pathways to the end-users. Few CLECs can, in a short period of time, afford the cost of duplicating the pathways to the end user which took decades for the ILECs to develop. Further, this would prove inefficient. The Telecom Act of 96, therefore, was intended to ensure fair, consistent, access to the installed end-user base of the ILECs by the CLECs (Competitive Local Exchange Courier). This meant that the ILEC would have to allow any CLEC access to their network. This was mandated without definition of the physical and logical interfaces between Telcos. Several aspects of the network, such as user and facilities databases, considered proprietary by the ILECs were not clearly defined as being part of the act. Softswitches Softswitches, are a newer mechanism for large-scale VOIP call routing. Softswitches get their name from the fact that traditional, special-purpose, Big Iron PBXs are not used route calls. Instead software-based call processing running on off-the-shelf hardware is used to perform the same function. Softswitches are used in conjunction with media gateways to interface VOIP calls with the Public Switched Telephone Network (PSTN). This is model very scalable in that in order to add capacity, only the off-the-shelf hardware need be upgraded. In some cases a VOIP provider may interface directly with another VOIP provider (without conversion to analog). In these cases, an IP-to-IP gateway is used to serve as the demarcation between the respective provider networks. There are several softswitch manufactures including Cisco Systems, BroadSoft, Sonus, Cilantro, and Lucent. These are considered class IV or class V switches. There are several call switching protocols unique to Telcos which are the centerpiece for switching calls on a large scale. One such protocol is Signaling System 7, which is essentially a global standard for call control. It is supported by every major PBX vendor in existence. It is the protocol, by which class IV and class V switches communicate. So a challenge faced by softswitches is that support for older Telco protocols is required while still supporting newer and emerging VOIP protocols. H.323 has been in existence for some time and is an older protocol for providing small-to medium scale VOIP call control. H.323 uses a peer to peer model, in that a call setup device is typically a peer to the device receiving the call. In some cases a this model may be expanded to a more hierarchical model by adding a Gatekeeper to aid in scalability. Media Gateway Control Protocol (MGCP) is newer VOIP protocol for the control of VOIP gateways. MGCP is a derivative of the older SGCP. This protocol does not, in itself, provide call control but is used to control gateways which serve as an interface to the PSTN.
7 Session Initiation Protocol (SIP) SIP is emerging as the protocol of choice to provide the greatest level of interoperability among disparate types of end-devices and disparate types of networks. The promise of SIP is that a vast array of devices will support it. This means that communication messages intended for a given individual will find its way to that individual regardless of the media used to communicate with that individual. This could include a voice call to a cell or PDA/cell phone combination, it could include for that individual, it could text messaging or a page to that person. In current reality, in the context of VOIP, SIP is an agreed upon standard that most VOIP manufactures support, in a least a minimal fashion to complete phone calls. In some manufactures implementation of SIP, only a subset of phone features are supported, as compared to other protocols. Minimally, one SIP phone can call another. Features, such as call-hold, call transfer, music-on-hold may not interoperate from one vendor s SIP implementation to another s. This is, in part due to the fact that SIP itself is intended to be extensible to the technology to which it is being applied. That is, there was no specific telephony extensions specified to the initial draft of SIP. These are being developed as the implementations are rolled-out. Multi Protocol Label Switching (MPLS) Multi Protocol Label Switching made great inroads in supplanting legacy Telco data transport technologies such as ATM. MPLS is described as a layer-3 control plane steering a layer-2 transport plane. Most legacy transport technologies operate at layer-2. These include ATM, Frame-Relay, Point-to-Point protocol, and the older X.25. MPLS allows native use of layer-3 routing, with which most data networking personnel are familiar. Given that MPLS supports native use of layer-3 routing, DSCP may also be conveyed, thus allowing the Telco to offer data services with different tiered pricing, based on the classification of service. This allows consumers to garner end-to-end guarantees for their high-priority traffic, while paying less of a premium for their low priority traffic. An example of an all-mpls service provider is Masergy. Their network was built from the ground up as an MPLS network. Most if not all other service providers have retrofitted at least portions of their network to include MPLS. Most equipment vendors include MPLS capable equipment in their product lines. The solution to aging Telco networks the addition of Softswitch and MPLS capabilities. Most if not all Telcos recognize this and are scrambling to modernize their network to include these capabilities. The solution to interoperability is to develop a set of open, standardized set or protocols and extensive third-party or open forum testing. To a great extent, the features offered by a Telco come at the request of its customers. Any Telco would be glad to deploy services or features they are sure the consumer will pay for.
8 E911 Emergency services have been an institution in modern society in North America, for several decades. Indeed, young children are taught when to use those three magic numbers and have become heroes for saving a life by following emergency service dispatcher s instructions. Life and property are saved when this service is used correctly. Conversely, life and property are lost when this service does not function properly. In the past and in the TDM paradigm, a specific hand-set, and more importantly the handset s location was known to be on the far-end of a specific set of copper wires connected to a PBX within a Telco s Central Office (CO). This situation could only be changed with the intervention of the Telco who owned the copper. Adding to the complexity, a number of Cable TV providers have made a foray into the local exchange market by providing dial-tone over the copper cable infrastructure they have in place. This has served to increase the number of CLECs. This hard-wired paradigm has changed with the advent of more convenient, personalized, and mobile communication systems. VOIP and wireless have presented challenges to the traditional Emergency Services infrastructure, largely due limitations imposed by the technology used to coordinate these services. Today s E911 infrastructure, in most cases, makes use of Centralized Automatic Messaging Accounting (CAMA) trunks co-located in each Telco CO serviced by the Public Service Access Point (PSAP). In some cases, this CO to PSAP connection is provided by a Primary Rate ISDN (PRI) interface. CAMA trunk technology has been in existence for several decades and has not been substantially modernized because it is in place, it works, and cannot be taken out of service without significant cost and consequence. Further, CAMA offers benefits not found in PRIs. In many cases, end-user 911 service is provided by a PRI. In some cases a CAMA trunk from an end-user The local PSAP servicing a community maintains the authoritative database of phone numbers to physical address mappings. This database is not accessible by the general public and is sometimes not updated when changes occur. Such an oversight may occur when an individual changes telephone service provider from an incumbent provider to a Competitive LEC. The Incumbent provider is no longer legally responsible to update the PSAPs database for that individual. The CLEC may neglect to do so, or may be acquired, or go out of business prior to completing such an installation. Further complicating the issue is the recent legislation regarding Local Number Portability. This legislation has afforded the consumer the freedom to take their phone number with them when they change telephone service providers. This legislation has, in
9 effect, served to force Telcos to surrender portions of their unified dialing-plan to other Telcos in an uncoordinated, piecemeal fashion. The inherent check of a PSAP knowing what number block, belongs to which Telco has been removed. Clearly the freedom to keep one s phone number comes with the risk of emergency services not being able to find that consumer. Also complicating the issue is that PSAPs are administered at the county level. Clearly, different counties have different priorities and budgetary expenditures toward E911 service are not consistent, nor necessarily commensurate with population. One of the solutions used by today s wireless carriers is to imbed location transceivers within the handsets. Then a handset may be located by GPS or triangulation of nearby cell towers. This is effective because the cost to produce sophisticated handsets has gone down and a cell phone is a personal item which one carries with them. This does not translate directly to VOIP because a PC may be an endpoint, and a person may wish to make VOIP calls from multiple PCs. The network cannot inherently know where your PC is at any give time. With wireless, Mobile IP, VPN, and GRE tunneling, even a specific IP address does not divulge geographic location. End-to-end IP connectivity is the only requirement for VOIP endpoints to make phone calls. Clearly, this aspect of VOIP is a double-edged sword. Unfortunately, looking to other countries dose not provide a clue to the resolution to this issue as few countries have as robust emergency services infrastructure, and those who do are grappling with the same issue. One near term solution to the overall problem is requesting that VOIP users sign-in and update their whereabouts through a web portal. Obviously, this has the drawback of placing the onus on the VOIP user, who may forget to sign-in or think that it is not important. Several VOIP providers are using this approach such as Vonage and RedSky. On a much smaller scale, assuming PSAP database is somewhat correct, products such as Cisco Systems Emergency Responder and IP Celerate can provide the exact location and extension of a VOIP phone which dialed 911. This is done through the use of Cisco Discovery Protocol and SNMP. The IPCelerate product can also alert campus security personnel who can typically respond in less time then emergency personnel who would not be familiar with the site. This approach is effective in a multi-story building or in a campus environment or perhaps even a small municipality, but does not solve the larger issue. The solution to this issue on a large scale has to be developed from a combination of elements including technical innovation, and consistent, detailed legislation.
10 Conclusion This document has examined several challenges faced by the designers and implementers of VOIP and some potential resolutions to those issues. Some significant issues faced include ensuring QOS, interoperability between Telcos, and maintaining E911 service to mobile VOIP users. In the final analysis, VOIP obviously provides cost and feature benefits which outweigh the challenges seen here and will continue to be direction of choice for both the Telco service providers and the consumer. It is incumbent upon the user communities, to ensure that VOIP is deployed in such a manner as to retain the benefits unique to it, while still retaining required functionality provided previously, by legacy systems.
5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5.1 LEGACY INTEGRATION In most cases, enterprises own legacy PBX systems,
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Implementing VoIP support in a VSAT network based on SoftSwitch integration Abstract Satellite communications based on geo-synchronous satellites are characterized by a large delay, and high cost of resources.
A by NEC Unified Solutions, Inc. Planning an IP Network for Voice and Data Contents 1 Introduction...3 2 Voice Quality Requirements...3 3 Codecs...3 4 Network Layout...5 5 Planning Capacity...6 5.1 Step
NEWT Managed PBX A Secure VoIP Architecture Providing Carrier Grade Service This document describes the benefits of the NEWT Digital PBX solution with respect to features, hardware partners, architecture,
WAN Technology Heng Sovannarith email@example.com Introduction A WAN is a data communications network that covers a relatively broad geographic area and often uses transmission facilities provided
52-30-15 DATA COMMUNICATIONS MANAGEMENT VOICE AND DATA NETWORK INTEGRATION Larry Schessel INSIDE Voice and Data Networks; Voice Over Internet Protocol; VoIP Products; Voice and Data Network Integration;
Agilent Technologies Performing Pre-VoIP Network Assessments Application Note 1402 Issues with VoIP Network Performance Voice is more than just an IP network application. It is a fundamental business and
VitalPBX Hosted Voice That Works For You Vital Voice & Data s VitalPBX VVD Hosted PBX solutions provide you with the stability of a traditional telephone PBX system and the flexibility that only a next
An Allstream White Paper Best Practices for deploying unified communications together with SIP trunking connectivity An Allstream White Paper 1 Table of contents Correct identification and maximization
Voice over IP Abdus Salam ICTP, February 2004 School on Digital Radio Communications for Research and Training in Developing Countries Ermanno Pietrosemoli Latin American Networking School (Fundación EsLaRed)
CTS2134 Introduction to Networking Module 07: Wide Area Networks WAN cloud Central Office (CO) Local loop WAN components Demarcation point (demarc) Consumer Premises Equipment (CPE) Channel Service Unit/Data
SIP Trunking DEEP DIVE: The Service Provider Larry Keefer, AT&T Consulting UC Practice Director August 12, 2014 2014 AT&T Intellectual Property. All rights reserved. AT&T, the AT&T logo and all other AT&T
Introduction Computer Network. Interconnected collection of autonomous computers that are able to exchange information No master/slave relationship between the computers in the network Data Communications.
TECHNICAL CHALLENGES OF VoIP BYPASS Presented by Monica Cultrera VP Software Development Bitek International Inc 23 rd TELELCOMMUNICATION CONFERENCE Agenda 1. Defining VoIP What is VoIP? How to establish
Contents VoIP Starts with SmartNode...3 Why SmartNode?...3 SmartNode Product Comparison...5 VoIP Appliance with Embedded Windows...7 Carrier-Grade TDM + VoIP SmartMedia Gateways...8 Enterprise Solutions...9