HD Voice: High-impact Strategies - What You Need to Know: Definitions, Adoptions, Impact, Benefits, Maturity, Vendors

Kevin Roebuck

Description

The Knowledge Solution. Stop Searching, Stand Out and Pay Off. The #1 ALL ENCOMPASSING Guide to HD Voice.

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""Here's Your Chance To Skip The Struggle and Master HD Voice, With the Least Amount of Effort, In 2 Days Or Less...""

Wideband audio is an audio technology used in telephony. It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 80 Hz to 14 kHz. Traditional, or narrowband telephone calls, limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio increases the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz or higher.

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This book is your ultimate resource for HD Voice. Here you will find the most up-to-date information, analysis, background and everything you need to know.

In easy to read chapters, with extensive references and links to get you to know all there is to know about HD Voice right away. A quick look inside: Wideband audio, Anti-tromboning, Back-to-back user agent, Call Agent, Call origination, Chatter bug, Comfort noise, Hookflash, Host Media Processing, International gateway, Internet telephony service provider, IP Multimedia Subsystem, Lawful interception, Media Gateway Control Protocol, Media Gateway Control Protocol (MGCP), Multimedia Telephony, Origination, Packet loss concealment, PacketCable, Purple minutes, Real-time Transport Protocol, Session border controller, Session Initiation Protocol, Signaling gateway, SIP connection, SIP Trunking, Softphone, STUN, Survivable Branch Appliance, Talkspurt, Traversal Using Relay NAT, Trunking gateway, Vishing, Voice chat, Voice engine, Voice peering, VoIP spam, VoIP VPN, WebCall, White, black and grey routes...and Much, Much More!

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Sujets

• Computers
• Informatique
• General officer
• General

Informations

• Publié par: Emereo Publishing
• Date de parution: 24 octobre 2012
• Nombre de lectures: 0
• EAN13: 9781743445075
• Langue: English
• Poids de l'ouvrage: 2 Mo

Extrait

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Sources, licenses and contributors of the articles and images are listed in the section entitled “References”. Parts of the books may be licensed under the GNU Free Documentation License. A copy of this license is included in the section entitled “GNU Free Documentation License”

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Contents

Articles Wideband audio Anti-tromboning Back-to-back user agent Call Agent

Call origination Chatter bug Comfort noise Hookflash Host Media Processing International gateway Internet telephony service provider IP Multimedia Subsystem Lawful interception Media Gateway Control Protocol Media Gateway Control Protocol (MGCP) Multimedia Telephony Origination Packet loss concealment PacketCable Purple minutes Real-time Transport Protocol Session border controller Session Initiation Protocol Signaling gateway SIP connection SIP Trunking Softphone STUN Survivable Branch Appliance Talkspurt Traversal Using Relay NAT Trunking gateway Vishing Voice chat

1 3 4 5 7 7 8 9 9 10 10 11 21 25 26 29 30 30 31 34 34 38 43 48 49 50 51 52 55 55 55 57 57 58

Voice engine Voice peering VoIP spam VoIP VPN WebCall White, black and grey routes

References Article Sources and Contributors Image Sources, Licenses and Contributors

Article Licenses License

59 60 60 61 62 63

64 66

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Wideband audio

Wideband audio

Wideband audiois an audio technology used in telephony. It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 80 Hz to 14 kHz. Traditional, or narrowband telephone calls, limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio increases the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz or higher. In 1987, the International Telecommunication Union (ITU) standardized a version of wideband audio as G.722. Radio broadcasters began using G.722 over Integrated Services Digital Network (ISDN) to provide high-quality audio from remote locations, such as sports venues. The traditional telephone network (PSTN) is generally limited to narrowband audio by the intrinsic nature of its transmission technology, TDM (Time-Division Multiplexing), and by the analogue-to-digital converters used at the edge of the network, as well as the speakers, microphones and other elements in the endpoints themselves. Wideband audio has been broadly deployed in conjunction with video conferencing. Providers of this technology quickly discovered that despite the explicit emphasis on video transmission, the quality of the participant experience was significantly influenced by the fidelity of the associated audio signal. Communications via Voice over Internet Protocol (VoIP) can readily employ wideband audio. When PC-to-PC calls are placed via VoIP services, such as Skype, and the participants use a high-quality headset, the resulting call quality can be noticeably superior to conventional PSTN calls. Some of the handset manufactured by Nokia which run S60 [1] and Series40 OS that support VoIP also support wideband audio . A number of audio codecs have emerged to support these services, supplementing G.722. Manufacturers of audio conferencing equipment have introduced wideband-capable models that include support for G.722 over VoIP. Conference calls are a direct beneficiary of the enhancements offered by wideband audio. Participants often struggle to figure out who is talking, or to understand accented speakers. Misunderstandings are commonplace due primarily to generally poor audio quality and an accumulation of background noise. Some listener benefits cited of wideband audio compared to traditional (narrowband):  Clearer overall sound quality  Easier to recognize voices, distinguish confusing sounds and understand accented speakers  Ease of deciphering words that have the close sounds o‘fs’and‘f’and others, often indistinguishable over telephone lines.  Ability to hear faint talkers and to understand double-talk (when more than one person is speaking at the same time)  Reduced listening effort (decreased cognitive load), resulting in increased productivity and lessened listener fatigue  Better understanding in the face of other impairments, such as when talkers are using a speakerphone or in the presence of background noise Despite its reputation for poor audio quality, the mobile telephone industry has started to make some progress on wideband audio. The 3GPP standards group has designated G.722.2 as its wideband codec and calls it Advanced Multirate–Wideband (AMR-WB). A few handsets have been produced supporting this codec (for example, Nokia), and network demonstrations have been conducted. After years of trials, AMR-WB was made commercially available in September 2009, when Orange launched the world's first high-definition voice service for mobile phones in Moldova, the first time since the 1990s that mobile voice technologies have been subject to a significant evolution. The first mobile handset integrating high-definition voice capability launched by Orange Moldova is the Nokia 6720c, which integrates WB-AMR. Orange has since

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Wideband audio

rolled out HD voice in the UK, France, Belgium, Romania, Armenia, Dominican Republic and Catalonia, with [2] Switzerland and Luxembourg due to follow.

Deployment As business telephone systems have adopted VoIP technology, support for wideband audio has grown rapidly. Telephone sets from Avaya, Cisco, Grandstream, Gigaset, Polycom (which brands wideband audio "HD Voice"), Snom, AudioCodes (which brands wideband audio "HDVoIP") and others now incorporate G.722, as well as varying degrees of higher-quality audio components. Suppliers of integrated circuits for telephony equipment, including Broadcom, Infineon, and Texas Instruments, include wideband audio in their feature portfolios. There are audio conferencing service providers that support wideband connections from these and other VoIP endpoints, while also permitting PSTN participants to join the [3] conference in narrowband. sipXtapi is an open source solution for VoIP media processing engine supporting wideband and HD voice that provides RTP and codecs through a plugin framework for use with SIP and other VoIP protocols. Skype uses an audio codec called Silk_(codec) which allows for extremely high quality audio.

A number of carriers around the world have rolled out HD voice services based on the G.722 wideband standard. In North America, hosted service providers have recently deployed the Aastra Hi-Q upgrade to its installed user base and as of January 2010 claimed around 70,000 HD voice endpoints. Consumer service provider ooma has an estimated 25,000 HD voice endpoints deployed stemming from its roll out of its second-generation Telo hardware. [4] Three UK launched HD voice calls onto their network at the end of May 2011 .

References [1] Forum Nokia - VoIP details (http:/ /wiki.forum.nokia.com/index.php/VoIP_support_in_Nokia_devices#Support_in_Series_40_devices) [2] http://event.orange.com/default/EN/all/hd_voice_en/hd_voice_country_availibility.htm [3] http://sipxtapi.sipfoundry.org/wiki/SipXtapi [4] http://blog.three.co.uk/2011/06/17/hd-voice-calls-launched-on-three/

External links  (http://www.hdvoicenews.com) HD Voice News  The Effect of Bandwidth on Speech Intelligibility (http://www.polycom.com/common/documents/ whitepapers/effect_of_bandwidth_on_speech_intelligibility_2.pdf) Polycom technical White Paper  TMCNet.com HD Voice Community (http://hdvoice.tmcnet.com/)  VoIP transitioning to High Definition Voice (http://weblog.infoworld.com/realitycheck/archives/2007/11/ voip_transition.html) InfoWorld blog on wideband audio  Texas Instruments HD Audio website (http://focus.ti.com/download/bcg/HDvoice.html)  Can You Hear What I Mean? Polycom Delivers HD Voice (http://www.polycom.com/common/documents/ whitepapers/can_you_hear_what_i_mean_polycom_delivers_hd_voice.pdf) Sponsored IDC White Paper  International Telecommunications Union  HD VoIP Sounds Better (http://www.audiocodes.com/Data/Uploads/HD VoIP White paper.pdf) AudioCodes White Paper  HD Voice Cookbook (http://www.wirevolution.com/hd-voice-cookbook/)

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Anti-tromboning

Anti-tromboning

Anti-tromboning(also referred to asanti-hairpinningormedia release) is a feature employed in telecommunication networks, such as voice over IP networks, that optimises the use of the access network and reduces excess processing and traffic. A network border node, such as a Session Border Controller, handling calls as they pass from the access network to the core network can examine the IP address or domains of both the caller and called parties and if they reside in the same part of the network the media path can be [1] released allowing media to flow directly between the two parties without entering the access network. The benefits of this action are twofold: 1) the caller is not paying for any bandwidth usage on the carrier network (but may be arbitrarily paying for the carrier's handoff service) and 2) The carrier's network is less congested.

In mobile networks servicing a large number of geographically dense peers, any two peers who wish to communicate between one-another may exchange their media data through a separate path that exploits localised and lower-power transmission, as a form of sub-band signalling. This extends into situations where bulk data can be sent over a less reliable but higher bandwidth capacity and cheaper cost/latency link, whilst parity data for reconstructing bad packets or supporting determinancy in fuzzy-state weakly determined data can be sent over more reliable but lower bandwidth capacity and expensive cost/latency link. The session and control data can be completely decentralised, removing the Tromboned line altogether, under a suitable multiple-input multiple-output MIMO network system. In this case, the aggregate media and session control data may be distributed across the network in a dynamic best-possible solution form that meets the link criteria of both the individual peer and the constraints of the network infrastructure (other peers, basestations, etc.), which may vary between each individual peers. A subscription service may primarily support paying members, but allocate a certain amount of under-utilised bandwidth to provide longhaul access to non-paying members, with the assumption that these members will in turn provide paying members with traversal of bulk media data within a geographical area, or even high-latency propagation across cells. The lowered cost to the service provider that results from resource sharing is the economic rationale for these network provision schemes.

References [1] Aoun et al. (2002-02). "Identifying intra-realm calls and Avoiding media tromboning" (http:/ /tools.ietf.org/html/ draft-aoun-midcom-intrarealmcalls-00). . Retrieved 2011-05-20.

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Back-to-back user agent

Back-to-back user agent

Aback-to-back user agent(B2BUA) is a logical network element in Session Initiation Protocol (SIP) [1] applications. SIP is a signaling protocol to manage multimedia Voice over Internet Protocol (VoIP) telephone calls. A back-to-back user agent operates between both end points of a phone call or communications session and divides the communication channel into two call legs and mediates all SIP signaling between both ends of the call, from call establishment to termination. As all control messages for each call flow through the B2BUA, a service provider may implement value-added features available during the call. In the originating call leg the B2BUA acts as auser agent server(UAS) and processes the request as auser agent client(UAC) to the destination end, handling the signaling between end points back-to-back. A B2BUA maintains complete state for the calls it handles. Each side of a B2BUA operates as a standard SIP network element as specified in RFC 3261. A B2BUA may provide the following functions: • call management (billing, automatic call disconnection, call transfer, etc.) • network interworking (perhaps with protocol adaptation) • hiding of network internals (private addresses, network topology, etc.) Often, B2BUAs are implemented in media gateways to also bridge the media streams for full control over the session. A signaling gateway, part of a session border controller, is an example of a B2BUA.

References [1] RFC 3261,SIP: Session Initiation Protocol, IETF, The Internet Society (2002)

External links • Open Source Sippy SIP B2BUA (http://www.b2bua.org)

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