WHITE PAPER. Specialized Hardware Answers Booming VoIP Transcoding Demands

Transcription

1 WHITE PAPER Specialized Hardware Answers Booming VoIP Transcoding Demands

2 Contents TRANSCODING: AN INTRODUCTION...3 WHAT IS TRANSCODING...4 WHY REAL-TIME TRANSCODING...4 WHEN TO USE TRANSCODING...5 AUDIO CODECS...6 COMPARING CODECS...6 DIFFERENTIATION FACTORS...6 VOICE QUALITY...8 TRANSCODING TRADEOFFS: SOFTWARE VS. HARDWARE...9 SOFTWARE TRANSCODING...9 HARDWARE TRANSCODING...9 APPLICATIONS AND USE CASES...10 TRANSCODING BENCHMARKING TEST RESULTS...10 SANGOMA TRANSCODING PRODUCTS CONCLUSION...14 ROYALTIES APPENDIX A /2011

3 Transcoding: An Introduction Transcoding is a growing requirement of today s disparate communications network environment. In a time when phone calls must often traverse multiple networks, and in which different networks often support different codecs and protocols, transcoding voice calls between various networks and end-point devices is increasingly necessary. This white paper explains transcoding and codecs, outlining their operation and application to VoIP, along with relevant products from Sangoma Technologies, delineating their value proposition and benefits for buyers. 3

4 What is Transcoding Generally speaking, transcoding is the conversion of data from one form or format to another. VoIP transcoding specifically is the conversion between one digital representation of voice, or codec, and another. It is required when two communicating IP-based phones, or end-point devices, have no codec in common that they both support. The transcoding of voice is analogous to the more widely experienced conversion of music from a standard CD (compact disc) format like.cda, into the more compact MP3 file format for playback in ipods and other similar consumer devices, in which the music retains most of its sound qualities while its digital representation, formatting and storage requirements are changed. However, the transcoding of voice calls is particularly demanding because it must take place in real-time, supporting instantaneous requirements of intelligible human conversation in which delays in the communication (commonly referred to as latency ) as short as 150 milliseconds, which can impede clear communication and make a phone conversation difficult, if not impossible to conduct. Why Real-Time Transcoding There are a number of conditions when real-time transcoding becomes necessary: 1. End-points want to participate in a communication session, even though they do not support the media format demanded by the session 2. Bandwidth, included radio spectrum, are a limited resource within a communications infrastructure Moving from a software-based transcoding to hardware-based transcoding solution becomes necessary as a response to computing resource limitations, where the primary resource limitation involves computing resources, such as the CPU and RAM-based memory. 4

5 When to Use Transcoding Transcoding is frequently required under a variety of circumstances including, for example:»» When either a receiving device or overall transmission environment (possibly due to bandwidth limitations) cannot support an initial format well»» To convert incompatible or outdated data formats to compatible, better-supported ones Transcoding can also commonly be required for applications that traverse multiple networks, prominently including international calling, extensive organizational networks, and Open Source and wireless applications. DIAGRAM 1: Transcoding for hosted VoIP services over broadband DSL example IP Asterisk Server G.711 G.729 LAN Router IP DSL Broadband Router LAN IP Phones D100 Central Office/ HQ Customer Premises Transcoding is also typically required for call recording, tone detection, and play announcements or tones. Besides multiple network traversals, another key indication for transcoding requirements is bandwidth scarcity, such as in cases of wireless interconnection, upstream (outgoing) DSL and some international environments. At times also, for example, a remote end-point may use G.729 or another high-compression codec to save bandwidth while the network backbone is using G.711, thus requiring transcoding. With increasingly complex interconnection between networks, flexible transcoding functionality is increasingly essential. Especially during the current period of industry-wide transition to all-ip networks, VoIP providers, like many others, must inevitably route calls across multiple networks and cross multiple IP network borders; increasing transcoding requirements. 5

6 Audio Codecs Transcoding is performed by computer programs (and/or devices incorporating such programs) that implement complex standards-based mathematical algorithms that automatically translate one data formatting system, or codec, into another. The term codec is actually a combination of the words coder-decoder since conversion is bi-directional, involving at once coding and decoding, with codecs combining functionalities of both a coder that encodes and compresses data representing voice for network transmission, and a decoder, operating simultaneously in both directions. Voice transcoding is typically a two-step process in which data representing voice in an initial format is decoded to an intermediate uncompressed format typically one known as PCM (pulse code modulation) and then encoded into its target, or final, format. Audio codecs used for VoIP typically apply algorithms that reduce the number of packets and the number of digital bits required to represent the audio. Besides voice, codecs are also prominently used for video and other media types. IP-based telephones contact each other during call set-up to arrive at common codecs through a capabilities exchange assessing which, if any, codecs are shared between them, then implementing selection rules between available alternatives. Wireless phones can typically support several different voice codecs. The ITU (International Telecommunications Union), the longstanding organization through which governments and companies coordinate global communications, is responsible for worldwide telecom codec standardization. Comparing Codecs Hundreds of codecs are currently in use but a few are particularly widespread. In ITU-speak, the codec known as G.711, u-law version, is the most ubiquitous in North America, especially as it is the normative codec for traditional circuit-switched telephony while important also for VoIP. Various versions of high-compression codecs G.729 (e.g., including G.729a) and G.723, among others, are also widely used for VoIP, along with newer wideband codecs, such as G.719 and G.722. Differentiation Factors Codecs typically vary by: Bit rate Required audio bandwidth Required processing power and memory Latency and loss characteristics Types of audio well-supported by each Resilience 6

7 One of the most important dimensions on which codecs differ is compression, the degree to which they reduce the volume of original data (in packets, or bits) to lessen the bandwidth required for transmission. There is often some tradeoff between high compression (and thus, lower bandwidth usage) and optimal voice quality. G.711, the longstanding standard for landline, toll quality voice, uses minimal compression. By contrast, for example, G.729 and G.723 dramatically reduce the number of packets and bits needed through advanced algorithms that model voice in a more compressed manner, using less bandwidth and so able to transmit more channels within given bandwidth constraints. G.711 may use three to five times as much bandwidth to send the same conversation as such high-compression codecs. G.729 is considered a high-complexity algorithm and G.729a is a somewhat less complex variant. Both are considered relatively high-quality for high-compression codecs and are popular for teleconferencing and visual telephony as well as VoIP and wireless applications while using much less bandwidth than G.711. Wideband or high-definition VoIP codecs are relatively recent introductions which re-expand VoIP telephony to 20kHz sound, meanwhile include L256, G.719 and the G.722 family. Table 1 below compares several audio codecs commonly used for VoIP across a variety of quantitative variables. TABLE 1: Basic Dimensions Of Common Voip Codecs Codec Bit Rate (Kbps) Sample Time (ms) Voice Payload (bytes) Overhead (bytes) RTP UDP IP Packet Size (bytes) Packets per Sec BW Reqs (Kbps) G G G G G G ilbc mode ilbc mode Source = Sangoma Technologies Corp. High-compression, lower-bandwidth codecs may in some cases offer voice quality comparable to what is considered normal on cell phones (though some do better). They may also add a slight additional delay reflecting incremental computational effort. High-complexity codecs also typically increase the costs of VoIP phones, 7

8 requiring faster and thus more costly processors as well as more memory. Related issues increase with phones that handle multi-party bridged calls. Early VoIP carriers typically standardized their networks on G.711, like the PSTN (Public Switched Telephone Network). The major disadvantage was that this codec cannot use the potential for lower bandwidth and thus higher channel density realized by other codecs, as well as the ability for improved ( wideband ) voice quality facilitated by some others. Nor does it provide flexibility in the face of variable bandwidth requirements. Voice Quality Besides compression, key factors affecting VoIP audio quality include: Delay (especially exceeding 100 milliseconds) Jitter (delay variation between packets) Lost and/or damaged packets Attention to network architecture, along with equipment choice and configuration, can reduce loss and delay and so therefore support voice quality. These include measures such as reducing the numbers of hops, bandwidth connections and router queue sizes. Private data networks (as opposed to Public Internet-based transmission) are meanwhile widely considered a requirement for high QoS to maintain VoIP quality. 8

9 Transcoding Tradeoffs: Software vs. Hardware Transcoding functionality originates in software but can be hardwired into standard (servers) or specialized computer hardware, the latter known as DSPs (digital signal processors), and integrated circuit devices optimized for complex algorithmic processing. Software Transcoding Software was the original form of transcoding technology. Transcoding can still be performed by host-based licensed software, which has the potential advantage of requiring smaller up-front incremental investment for smaller organizations as in a SOHO (Small Office/Home Office) environment with modest transcoding demands. With software transcoding, the host CPU performs the algorithms. The combination of Echo Cancellation with transcoding, both real-time digital signal processes, radically increases demands on CPU resources. Combined with the requirements of other customer applications, this creates intense competition for computing resources and runs up against capacity and scalability constraints, especially for larger organizations with significant application demands. This is illustrated in the benchmark test charts in Appendix A, which compare software transcoding with DSP hardware transcoding; demonstrating at least 50 percent system load reductions and 100 percent capacity increases with the latter. With the demands of real-time voice delivery in which delays of milliseconds can be problematic for conversation voice transcoding is thus an extremely processing-intensive activity that diverts computer resources on-demand from other activities if executed over shared infrastructure and so is likely to negatively affect other customer applications. Hardware Transcoding The ability to instead offload transcoding to specialized DSP hardware means reduced CPU loads that preserve system resources and stability. Where CPU resources are inadequate, transcoding can meanwhile also negatively affect voice quality to the extent that it introduces delay. While software can thus work well in small-scale applications, they are not adequate for large enterprise-level or service provider demands. Hardware-based transcoding, especially with DSPs, provides greatly increased capacity and scalability, along with broader codec support. Dedicated DSPs with codec capabilities avoid resource competition, more reliably bringing the large computing resources required for voice transcoding to bear in real time without sacrificing other vital applications. Transcoding is in a sense downloaded to hardware optimized for these tasks with reserves of highly specialized capacity 9

10 without using up precious host CPU resources. DSP-based solutions also offer cost, efficiency and voice quality benefits. Hardware has basic economic advantages over codec software in its ability to maximize revenue with economies of scale. They can serve the largest set of customers at a relatively attractive cost per channel, easing deployment while supporting a more comprehensive set of voice codecs for both wireline and wireless providers, from the customer premise or network edge. Hardware can meanwhile use defined service policies to assign specific codec treatment to calls on virtually any basis, including for example by IP address, trunk group or other attributes. Applications and Use Cases Common transcoding applications include situations that require traversal of multiple networks and/or connectivity between services and/or networks otherwise using different codecs. Prominent examples include:»» Multi-location enterprises and organizations, for example, connecting headquarters IP-PBXs to many branch offices across multiple networks nationally or internationally»» International calling including use of call arbitrage, also connecting many sites across multiple networks globally»» Mobile call termination to wireline phones, especially internationally, often requires transcoding between AMR, used on 2G and 3G cellular networks, and codecs used in the local landline environments»» Open Source Telephony: Transcoding is particularly important also for Open Source Telephony which often features incremental transcoding requirements between Open Source gear and proprietary equipment used to interconnect networks. In the Open Source environment, software provides PBX functionality more typically accomplished by hardware under Open Source licensing that lets others use and add to what is available»» Storage of audio and video content files, for which transcoding allows economical storage of one maximumresolution version of each file, from which transcoding equipment can derive any version required by various phones and other receiving devices. This obviates costly and inconvenient requirements to store an extensive inventory of multiple versions of each file to be compatible with a wide range of devices Transcoding Benchmarking Test Results Sangoma recently conducted an extensive transcoding benchmarking test comparing software- and hardwarebased transcoding between VoIP and TDM voice transmission using G.711 and, respectively, codecs G.729 and ilbc, and employing Open Source Asterisk as a VoIP gateway. Detailed results are included in Appendix A of this paper. 10

11 The test method was to progressively increase numbers of active calls generated six times per second and monitor CPU load until the system could no longer process more calls, repeating various scenarios using different server types. The test found hardware transcoding consistently delivered lower CPU occupancy, or utilization, compared with software-based solutions and that it therefore allowed integrators to: Increase telecom platform port density Produce more with similar investment Reduce expenditures Deliver greater system stability DIAGRAM 2: Generic Test Set-up LAN Asterisk Based Gateway Server G.711 Encoded Voice VoIP Encoded Voice 1 Traffic Originator Loop Back Trunks 4 A108 3 D100 5 VoIP Encoded Voice 2 VoIP Encoded Voice 6 Traffic Receiver G.711 Encoded Voice 1. Traffic Generator is a SIPp script, establishing VoIP SIP Calls 2. VoIP gateway receives calls 3. D100 Transcoding board decodes to G.711, sends calls to A108 Digital T1/E1 board 4. Call is looped back on A108 Digital board 5. Call re-encoded in by D100 Transcoding card 6. Gateway makes call on VoIP network to Traffic Receiver, also a SIPp script written to terminate VoIP SIP calls Repeat 1 to 6 at a rate of 6 calls per second until CPU of Gateway Server can no longer take calls. 11

12 Sangoma Transcoding Products Sangoma produces specialized, dedicated DSP (Digital Signal Processor)-based circuit boards called the D-Series Transcoding boards, and the new NetBorder Transcoding Gateway in a 1U form factor appliance. The company s D100, D150 and D500 models are available in multiple form factors. They support any-to-any codec combinations, each consistently lowering CPU occupancy compared with alternative solutions, and dramatically saving on costs for bandwidth and data center expenses, like electric power and rack space. Sangoma calculates that transcoding based on its high-end D500, compared with a similar volume of standard CPU-based transcoding, for example, as requiring roughly half the cost, a fifth of the space, and an eighth of the power and ongoing operating expense. At the same time, Sangoma supports the widest range of VoIP audio codecs in the industry. Table 2 below lists these supported codecs. TABLE 2: Codecs Supported by Sangoma D-Series Transcoding Cards Wireline Wireless G.711 GSM - FR G.722 GSM - EFR G (HD) AMR G AMR-WB (G.722.2) G.726 G.729 Source = Sangoma Technologies Corp. THE D-SERIES: VOICE TRANSCODING Available in PCI and PCI Express form factors, Sangoma s transcoding boards convert simultaneous sessions of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. For more information, visit: D100 PCI with 1U bracket NETBORDER TRANSCODING GATEWAY The NetBorder Transcoding Gateway (NTG) offers an elegant and simple solution, by providing integrated high density transcoding in a 1U form factor appliance. The NTG allows running from 400 to 4000 transcoding sessions of a wide range of narrow band and HD voice codecs with unmatched quality. Transcoding Appliance 12

13 ABOUT THE AUTHOR OF THIS WHITE PAPER Marc Robins is an internationally recognized authority in the field of IP telephony and emerging new IP communications technologies. Marc is the Chief Technology Evangelism Officer of RCG (Robins Consulting Group), an IP communications industry consultancy providing an array of marketing, research and advisory services. Marc also serves as the president and managing director of the SIP Forum, a leading IP communications industry association. Prior to founding RCG, Mr. Robins served as vice president of publications and trade shows, associate group publisher and group editorial director at TMC, where he helped launch publications such as Internet Telephony and served as chief architect and conference co-chairman of the ITEXPO trade shows. For more information about RCG, call or info@robinsconsult.com. This white paper was commissioned by Sangoma Technologies. ABOUT SANGOMA TECHNOLOGIES Sangoma is a leading provider of hardware and software components that enable or enhance IP Communications Systems for both telecom and datacom applications. Enterprises, SMBs and Carriers in over 150 countries rely on Sangoma s technology as part of their mission critical infrastructures. Through its worldwide network of Distribution Partners, Sangoma delivers the industry s best engineered, highest quality products, some of which carry the industry s first lifetime warranty. The product line in data and telecom boards for media and signal processing, as well as gateway appliances and software. Founded in 1984, Sangoma Technologies Corporation is publicly traded on the TSX Venture Exchange (TSX VENTURE: STC). Additional information on Sangoma can be found at 13

14 Conclusion Transcoding of VoIP calls between networks is a growing requirement of an increasingly complex multi-network, multi-technology environment with a continually expanding number of communications end-points. Specialized DSP-based hardware, rather than software or standard servers, is the most cost-effective response for larger organizations and service providers. Such solutions help avoid competition for host CPU resources and the likely resulting problems for other applications, as well as voice quality stemming from voice transcoding s extremely resource-intensive real-time requirements. At the same time it also offers major cost savings. This issue will inevitably grow in importance also as video, which often requires transcoding, becomes increasingly important in communications across distance. Sangoma Technologies has a long history of successfully offering a broad range of high-quality DSP-based products to carriers worldwide. Royalties Some codecs like the popular G.729 charge users royalties by the port. Sangoma Technologies integrates most royalty/licensing costs into its own hardware price, so customers pay a single cost without any future licensing concerns, providing a significant customer benefit. Please note that this does not apply to the wireless codec AMR. Voiceage (the patent pool for AMR) requires that the onus of paying royalties should be downstream i.e. near the user, instead of from technology component suppliers. This means that the last integrator that sells to end users a complete solution, such as a conferencing server or phone set that supports AMR is the entity responsible for paying AMR royalties, and since a DSP is not a complete solution, a board component such as those sold by Sangoma does not qualify and Sangoma cannot pay royalties to the patent pool. For more information about AMR licensing terms, please visit here: 14i

15 Appendix A BENCHMARK RESULTS 100% increase in capacity 50% decrease in CPU G.711 to G.729 Atom Server G.711 to G.729 Atom Server G.711 to G.729 Dual Core Server G.711 to G.729 Quad Core Server System Load System Load System Load 100% 75% 50% Software Codec Sangoma D100 Baseline (G.711 only) 50% system load reduction 100% capacity increase System Specifications: CPU: Intel Atom Dual Core 1.66 GHz 25% Memory: 2 GB OS: CentOS release 5.6 x86 Software: Asterisk configured as a media gateway 1 x A108 Sangoma T1/E1 board 0% 1 x D100 Sangoma Transcoding board RTP packet size: 20ms G.711 to G.729 Dual Core Server Number of Simultaneous Calls 100% 75% 50% Software Codec Sangoma D100 Baseline (G.711 only) 2011 Sangoma Proprietary 2011/08/ % system load reduction Over 100% capacity increase System Specifications: CPU: Pentium Dual Core 2.50 GHz 25% Memory: 2 GB OS: CentOS release 5.6 x86 Software: Asterisk configured as a media gateway 2 x A108 Sangoma T1/E1 board 0% 1 x D100 Sangoma Transcoding board RTP packet size: 20ms G.711 to G.729 Quad Core Server Number of Simultaneous Calls 100% 75% 50% 25% 0% 2011 Sangoma Proprietary 2011/08/30 2 Software Codec Sangoma D100 Baseline (G.711 only) Number of Simultaneous Calls 50% system load reduction Over 100% capacity increase System Specifications: CPU: Intel Core 2 Quad 2.33 GHz Memory: 4 GB OS: CentOS release 5.6 x86_64 Software: Asterisk configured as a media gateway 2 x A108 Sangoma T1/E1 board 1 x D100 Sangoma Transcoding board RTP packet size: 20ms 2011 Sangoma Proprietary 2011/08/