Distance Conferencing:
Distance Conferencing:
Introduction:
With markets becoming ever global and travel costs ever increasing, there has been a dramatic shift in recent years towards the use of distance conferencing by many private and state organizations. Meetings spaces such as Boardrooms, Training Rooms, Classrooms, Courtrooms, Conference Halls, L H ll L i l i Ch b d S i l O i
from the mic/speaker, the intelligibility of audio is acceptable, however add a few more attendees who are forced to sit 4ft from the mic/speaker, and they are actually hearing the caller 6dB quieter, and likewise the caller is hearing them 6dB quieter. In the table‐top conference phone example, for the attendees
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Lecture Halls, Legislative Chambers and even Surgical Operating Theaters are adopting the use of distance conference systems. The purpose of this Application Note is to clarify the different types of conferencing system commonly installed, and how the Q‐Sys hardware interconnects with each type.
who are now seated closer ‐ say 4ft from the mic/speaker, the intelligibility of audio is acceptable, however add a few more attendees who are forced to sit 8ft from the mic/speaker, and once again they are hearing the caller 6dB quieter, and the caller is hearing them 6dB quieter. The further away the attendee sits from the mic/speaker the more this problem
Tele‐conferencing Overview (live audio only):
At its most basic form, a teleconference system could be a simple desktop telephone set to “speaker” mode, thereby allowing two or more listeners seated around the “speaker” phone to participate in the conversation. However a typical
attendee sits from the mic/speaker, the more this problem escalates, and that’s without even factoring in ambient background noises such as projector fans, HVAC, rustling papers, squeaky chairs, and general murmur.
speaker‐phone would be completely unsuitable for a larger boardroom table, which has led many companies such as Polycom, Cisco, Avaya and ClearOne to provide dedicated table‐top (“starfish”) conference phones, with 360 degree microphone pickup coverage and slightly louder speaker
l b k H th d di t d t bl t f playback. However these dedicated table‐top conference phones still have their limitations….
These limitations are caused by a variety of acoustic factors, the most significant being inverse square law – this dictates that every doubling of distance will attenuate the sound by that every doubling of distance will attenuate the sound by
Web‐Conference Overview (live audio + shared desktop):
This is a very common use‐case application with many corporate entities, where a Host schedules a Web meeting or Webinar using a PC desktop application (ie Skype, WebEx, GoToMeeting, FuzeMeeting, TalkFusion, AnyMeeting, Live Meeting etc) and as part of the scheduling, the Host will invite l i l ll Th W b C f li i i multiple remote callers. The Web‐Conference application is primarily used by the Host to share their PC Desktop (Data) for the purposes of screening a presentation to the remote callers, while the audio can either be streamed over the Web using the PC computer’s internal mic/speaker, or through a dedicated table‐top teleconference phone which is also dialed into the For meeting rooms where the teleconference system is
required to reinforce a larger meeting space, a larger meeting attendance (or both), the only solution is to minimize the distance between the talker and the microphone, and this can
table top teleconference phone which is also dialed into the meeting.
This usually works well while the Presenter is located near the PC microphone or Tabletop phone, however when the presentation concludes and the meeting is opened up for only be achieved by adding multiple microphones around the
room. While this approach will dramatically improve the clarity and intelligibility of the talker to the far end listener, it must be pointed out that each individual microphone requires its own dedicated Acoustic Echo Cancelling (AEC) processor.
discussion among a larger group, the Web application is often expected to perform like a dedicated “conference” system, and once again this will suffer from the same acoustic limitations as the tele‐ and video‐conference applications described above!
Acoustic Echo occurs when the remote caller’s voice comes from the “far end” and is broadcast over loudspeakers into the “near end” room. The sound is picked up by microphones in the room and echoed back to the remote caller. More information on the Q‐Sys Acoustic Echo Canceller processing information on the Q Sys Acoustic Echo Canceller processing component can be found in the White Paper:
Video‐conferencing Overview (live audio + video + data):
At its most basic form, a videoconference system could be a simple desktop PC running Skype with a webcam and internal microphone, thereby allowing two or more listeners seated around the desktop PC to participate in the video conversation and even view a shared desktop screen. Unfortunately this
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consumer quality web video streaming is notoriously unreliable, suffering from regular dropouts and screen freezes, and therefore completely unacceptable in a high‐power boardroom situation. For uninterrupted video conferencing there is a small number of manufactures such as Polycom, Cisco and LifeSize that provide specialized video‐conference
Video‐Conference System
Cisco, and LifeSize that provide specialized video conference systems with a dedicated network server & gateway for streaming high definition (and virtually uninterrupted) audio & video, typically using the standard H.323 codec. Although very expensive, these dedicated video conference systems still suffer from the exact same acoustic limitations as the
Common Audio Transport Protocols:
There are a number of widely used “transport” protocols teleconference application described previously. Once again
the only real solution is to add more microphones around the room, with the understanding that each of these microphones requires its own dedicated Acoustic Echo Canceller (AEC).
available for transmitting audio to / from the remote caller (or what is referred to as the Far‐End) for distance conferencing. The chart below highlights the audio quality difference between some of these more popular protocols:
Type: Transmission Protocol: Audio: Video: Data: Audio Quality: Observations:
Tele‐conference POTS (Plain Old Telephone Service) √ Low 7kHz upper bandwidth Tele‐conference ISDN (Integrated Services Digital Network) √ Medium 14kHz upper bandwidth Tele‐conference VoIP (Voice over Internet Protocol) √ High 22kHz upper bandwidth Web‐conference TCP/IP (Transmission Control Protocol over IP) √ limited √ Low–medium Subject to interruptions Video‐conference RTP/IP (Realtime Transport Protocol over IP) – H.323 √ √ √ High Free of interruptions
Telephone Hybrid Overview
In teleconferencing, a telephone hybrid (or telephone balance unit ‐ TBU) is often employed as an impedance‐matching interface between the 600ohm telephone line (typically from an RJ11 connector carrying a 2‐wire mixed send/receive), and professional grade audio equipment. Think of it as a DI box for
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The Analog Hybrid – these are very simple circuits & therefore very low cost, however the audio performance is quite low, with typically 20dB or less of trans‐hybrid separation between the send & receive channels. These are not recommended for high quality Tele‐conferencing applications.
telephones. While these professional telephone hybrids were originally conceived for the Talk Radio & TV broadcast industry as a means to add outside calls to the live mix, the Hybrid has recently found its way into telephone conferencing as a means
to interface digital conference mixers with analog POTS lines. The Digital Hybrid – these are much more complex devices and very expensive By employing Digital Processing such as Line One major challenge with telephone hybrids is the Line Echo
that occurs when the mixed send & receive signal are “split out” from the 2‐wire RJ11 connector. With an analog circuit it is very difficult to achieve satisfactory separation, and when the send signal returns down the phone line it will often include
very expensive. By employing Digital Processing such as Line Echo Cancellation (not to be confused with Acoustic Echo Cancellation), a Digital Hybrid is able to achieve anywhere from 32‐50 dB of trans‐hybrid separation between the send & receive channels. Because these Digital Hybrids are designed for mix‐minus Broadcast applications, the DSP engine will often some of the original receive signal. This round‐trip signal path
will result in a diffuse‐sounding “Line Echo” at the far end. A good quality telephone hybrid should have high trans‐hybrid loss (32dB or higher).
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include additional processing features such as Automatic Gain Control, Ducking and Dynamic Equalization that are applied across the entire output and fed back to the caller (the far end). While these features may be useful in broadcast where there is only one near‐end microphone, in tele‐conferencing
li ti th l lti l i h d There are basically two types of Telephone Hybrids: applications there are always multiple microphones, and any
processing should to be applied to each individual microphone. Therefore when using a Digital Hybrid with a Digital Conference Mixer such as the Q‐Sys Core 250i or Core 500i, the Digital Hybrid’s dynamic processing should be disabled in order to take Hybrid s dynamic processing should be disabled in order to take full advantage of the Core’s ability to assign independent processing across individual microphone inputs.
VoIP Overview
VoIP stands for Voice over Internet Protocol, and is basically the future of telephony. Not only does VoIP have a wider audio frequency response (making it sound like the caller is actually in the room with you), but a VoIP‐to‐VoIP call also provides true separation between the Send/Receive channels, effectively
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conferencing. This feature is not dependent on hardware, but is instead implemented in software within each Q‐Sys Core, therefore it comes at no cost. Furthermore multiple Softphone components can be used within a single Q‐Sys Design, making it simple to create multi‐room systems within a single Core.
Th S f Ph ll Q S i IPBX
eliminating the line echo that plagues many analog systems. A growing number of corporate and state organizations have already changed over (or are in the process of changing over) their older analog PABX (Private Automatic Branch eXchange ‐ abbr. PBX) to a modern digital VoIP‐PBX (abbr. IPBX). If a Digital Conference Mixer has the capability of streaming VoIP directly
The SoftPhone component allows Q‐Sys to register to any IPBX (such as Cisco, Avaya, FreeSwitch, Asterisk, etc.) or register to external SIP Trunking providers. It is also possible to use the SoftPhone component in unregistered mode, which allows making ad‐hoc IP‐to‐IP calls and to connect to other non‐ registered SIP‐compatible equipment The ITU standard SIP Conference Mixer has the capability of streaming VoIP directly
onto the Local Area Network (LAN), then this will eliminate the need for any Hybrid hardware altogether.
If an organization is using an older analog PBX phone system and may not be upgrading until a future date, it is still very
registered SIP compatible equipment. The ITU standard SIP codecs supported are G.711 (µ ‐law and A‐law), G.722, G.726 and uncompressed PCM. These protocols are compatible with most known IPBX providers such as Avaya, Cisco and Shoretel.
Dialing Overview
simple to patch a VoIP endpoint into their analog PBX. Simply add a third‐party FXO (Foreign eXchange Office) gateway between your VoIP endpoint and any analog POTS line. An FXO gateway is a significantly cheaper option than using Hybrid hardware, and provides a future‐proof solution for when the i ti d t ll d id t d t di it l
Whether you employ the Telephone Hybrid method or the digital VoIP method, the one commonality between them is the Call Management function. This includes a touch‐tone dialer, using industry standard DTMF tones (Dual Tone, Multiple Frequency), where the touch‐tones are generated for dialing
t d d t t d h ll i di li i organization does eventually decide to upgrade to a digital
IPBX.
VoIP in Q‐Sys
Q‐Sys Designer software Ver3.1 and higher contains a Softphone component (or VoIP telephony endpoint), which
out, and detected when a caller is dialing in.
The Q‐Sys Softphone component’s dialer includes all standard functions such as DTMF, Caller ID, Redial, Adjustable Auto‐ answer, DnD (Do not Disturb) plus support for Entry/Exit tones. Softphone component (or VoIP telephony endpoint), which
Hybrid‐to‐PSTN (Public Switched Telephone Network)... (Outside line) Hybrid to PSTN (Public Switched Telephone Network)... (Outside line) Hybrid‐to‐PABX (Private Automatic Branch eXchange – abbr. PBX)... (Inside line)
Example: Telos Hx1
Telephone Hybrid
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VoIP‐to‐SIP Trunk Provider... (Outside line) VoIP‐to‐FXO‐to‐PSTN... (Outside line) VoIP to SIP Trunk Provider... (Outside line)
VoIP‐to‐VoIP/PBX ‐ abbr. IPBX... (Inside line)
VoIP to FXO to PSTN... (Outside line) VoIP‐to‐FXO‐to‐PABX... (Inside line)
Example: AudioCodes
MP-114 FXO Gatewayy
TelePresence H.323 Codec transmits High‐Resolution Audio, Host PC running a web application / video through the USB TelePresence H.323 Codec transmits High Resolution Audio,
Video, Camera Control & Data (Desktop sharing) Host PC running a web application / video through the USB
