question: Why is there always static/interference during hold music when the person on the other end is crystal clear?
comment: To transmit sound over a telephone connection most efficiently, it is encoded at one end and decoded at the other end. These encoders and decoders (I'll call them CODECS from now on) are designed to work very efficiently, using minimum processing power on your phone (saving battery) and using minimum amount of data ('bandwidth') on the mobile link. Whenever you design something to work efficiently at one thing, you make tradeoffs in another thing - a sports car isn't good at crossing rough ground, and an all-terrain vehicle isn't very fast. In this case, the CODEC is designed to be best at making speech understandable. It is optimised for the patterns, frequencies and tones of human speech: the person speaking is crystal clear. When you're put on hold, the system plays you music, which has very different patterns of tones, beats, and a wider frequency range, which the CODEC is not optimised for. It has to work 'harder', and will use more power or bandwidth if possible, or lower quality if not. There are may different CODECs, but most of them just play you static when they run out of bandwidth. This is because because experiments have found that in speech (which is what the CODEC is optimised for) our brains prefer static noise compared to the line going silent for a second. When the hold music is playing, the CODEC - which is designed for the different sounds of speech - is overworked and throws in static bursts for bits it can't do. It's not a deliberate conspiracy to make you hang up. TL:DR - phones connections are optimised for transmitting speech and rubbish at music.
passages: With the application of newer equipment, music on hold devices can now interact with the caller. No additional programs are required on the platform, as all the logic is done with the MoH device. This can include services such as 'polling on hold', rating your customer service officer anonymously, etc. 
  
  The trend toward hosted IP telephony for business phone systems is demanding changes in music on hold message technology.
  
  Section::::Legal aspects.
  
  Section::::Legal aspects.:Copyright law.
   These acoustic technologies must also remove or reduce the noise levels so that the caller is well understood. A person making a call from a handsfree device who is in a busy restaurant or while driving will introduce large levels of noise into the call. This situation is complicated as the software must not only remove the noise around him, but must transmit his voice clearly and loudly to whoever is connected to him. 
   Music on hold
  
  Music on hold (MOH) is the business practice of playing recorded music to fill the silence that would be heard by telephone callers who have been placed on hold. It is especially common in situations involving customer service.
  
  Music on hold is sometimes referred to as phone on hold, message on hold, on hold messaging, or hold music.
  
  Section::::Development.
   A CNN survey found that 70 percent of callers in the United States who hold the line in silence hang up within 60 seconds. Meanwhile, research by North American Telecom found callers hearing music on hold will stay on the line 30 seconds longer than callers experiencing silence.
  
  Section::::Source of music.
  
  Section::::Source of music.:Radio.
  
  The musical source may be as simple as playing a local radio station through the MOH jack. This is unlawful without the express permission of the music title and mechanical copyright holders, or their representatives.
   In telephony, a call may be placed on hold, in which case the connection is not terminated but no verbal communication is possible until the call is removed from hold by the same or another extension on the key telephone system. Music on hold or on hold messaging may be played for the caller while the call is on hold, especially if the call has been placed to a customer service center. Alternatives to placing a caller on hold include virtual hold or virtual queuing solutions that allow scheduled or queue-based callbacks to be made to the caller. 
   BULLET::::- Samsung YP-S MP3 player "pebble" uses USB-to-3.5 mm TRRS jack adapter for charging as well as for data transfer.
  
  BULLET::::- On CCTV cameras and video encoders, mono audio in (originating from a microphone in or near the camera) and mono audio out (destined to a speaker in or near the camera) are provided on one three-conductor connector, where one signal is on the tip conductor and the other is on the ring conductor.
   However, there are some minor benefits to driving the line with a fully differential output:
  
  BULLET::::- The electromagnetic field around a differential line is ideally zero, which reduces crosstalk "into" adjacent cables, useful for telephone pairs.
  
  BULLET::::- Though the signal level would not be changed due to nominal level standardization, the maximum output from the differential drivers is twice as much, giving 6 dB extra headroom.
   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 videoconferencing. 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.
   These equipment are typically at the edge of the core networks that acts as a gateway between mobile core networks (IP based) and digital PSTN networks. Mobile core IP networks carry voice encoded in the form of one of the GSM/UMTS codecs (e.g.: GSM AMR). When this has to be carried over a G.711/PCM based PSTN network, the gateway equipment transcodes from GSM/UMTS codec to G.711 PCM samples. This results in a certain loss of voice quality.
   BULLET::::1. Audio latency: Most musicians who play in a group rely on audio cues to maintain tempo and other communication. The most obvious example is the use of a drummer in pop music to keep time. In a normal rehearsal or performance environment, there is a delay between the time one musician plays and another hears the sound due to the speed of sound in air. This is typically from 3 - 50ms. When audio is transmitted through a digital medium (i.e. the Internet), the delay (or latency) can be much longer. Cell phones have latency of approximately 50ms. Applications such as Skype have approximately 100ms. And, QuickTime and Windows Media server-based streaming systems add 8 or more seconds of latency. When the latency is too high, the audio cues are no longer effective. Researchers have shown that some musicians can ignore this delay while others find it obtrusive. In general, the higher the audio quality and the more channels transmitted, the higher the latency needs to be to reliably transmit the audio stream.
   The most common situation in which a mix-minus is used is when a telephone hybrid is connected to a console, usually at a radio station. The person on the telephone hears all relevant feeds, usually an identical mix to that of the master bus, including the DJ's mic feed, except that the caller does not hear his own voice. Because the feed being returned to the telephone has an identical mix, minus only one input, it is most efficient to employ a mix-minus as opposed to creating and mixing for a new independent output bus..
   Modern SDR-based receivers can completely eliminate this by utilizing "brick-wall" filters narrower than the channel spacing that reduce signals outside the passband to inconsequential levels. Where such an overlap within the passband occurs, a high pitched whistle at precisely 9 or 10 kHz can be heard. This is particularly common at night when other carriers from adjacent channels are traveling long distances due to atmospheric bounce.
   In a video demonstration, Corning showed a ClearCurve drop cable being wrapped dozens of times around a small metal rod, and suffering almost no signal loss and providing a perfect video feed. A conventional cable wrapped around the same rod completely lost the signal after only two turns.
  
  Section::::ClearCurve.:FTTH uses.
   In packet radio systems such as GPRS/EDGE, it is possible to combine DTX with capacity increase when VoIP is used for telephony. In such cases, resources freed up when one user is in silence can be used to serve another user. The increase of the number of users will contribute to the interference level. Systems that use voice codecs such as AMR can reduce vocoder rate adaptively to better combat interference. 
   Music on hold was created by Alfred Levy, an inventor, factory owner, and entrepreneur. In 1962, Levy discovered a problem with the phone lines at his factory. He discovered that a loose wire was touching a metal girder on the building. This made the building a giant receiver so that the audio broadcast signal from a radio station next door would transmit through the loose wire and could be heard when calls were put on hold. Levy patented his work in 1966. While other advancements have come to change and enhance the technology, it was this initial patent creation that began the evolution for today's music on hold.
   Section::::Reception and legacy.
   The MCU consists of two logical components:
  
  BULLET::::1. A single multipoint controller (MC), and
  
  BULLET::::2. Multipoint Processors (MP), sometimes referred to as the mixer.
  
  The MC controls the conferencing while it is active on the signaling plane, which is simply where the system manages conferencing creation, endpoint signaling and in-conferencing controls. This component negotiates parameters with every endpoint in the network and controls conferencing resources.
   Some systems are capable of multipoint conferencing with no MCU, stand-alone, embedded or otherwise. These use a standards-based H.323 technique known as "decentralized multipoint", where each station in a multipoint call exchanges video and audio directly with the other stations with no central "manager" or other bottleneck. The advantages of this technique are that the video and audio will generally be of higher quality because they don't have to be relayed through a central point. Also, users can make ad-hoc multipoint calls without any concern for the availability or control of an MCU. This added convenience and quality comes at the expense of some increased network bandwidth, because every station must transmit to every other station directly.
   Acoustic echo cancellation (AEC) is a processing algorithm that uses the knowledge of audio output to monitor audio input and filter from it noises that echo back after some time delay. If unattended, these echoes can be re-amplified several times, leading to problems including:
  
  BULLET::::- The remote party hearing their own voice coming back at them (usually significantly delayed)
  
  BULLET::::1. Strong reverberation, which makes the voice channel useless
  
  BULLET::::2. Howling created by feedback
  
  Echo cancellation is a processor-intensive task that usually works over a narrow range of sound delays.
  
  Section::::Technology.:Bandwidth requirements.
   The cable harnesses used in sound reinforcement and recording studios are called "multicores", also known as "snakes" or "looms". They carry audio signals between a mixing console and stage box. Modern digital mixing consoles typically use a single twisted pair cable rather than a traditional analog multicore.
  
  Section::::Industry quality standard.
   Section::::Interference reduction.
  
  Balanced audio connections use a number of techniques to reduce noise. 
   Some users of the iPhone 4S reported the random appearance of echoes during phone calls made with earphones in the initial release of iOS 5. The other party in the call was sometimes unable to hear the conversation due to this problem.
  
  Section::::Reception.
   It is vital that ViLTE video calls are allocated appropriate quality of service (QoS) to differentiate and prioritize this delay and jitter sensitive conversational traffic from other streaming video traffic that is not as delay or jitter sensitive. The mechanism used is called QoS Class Identifier (QCI). The ViLTE bearer traffic is typically allocated QCI=2, and the SIP-based IMS signalling QCI=5.
  
  Section::::Devices.
   Note: the chosen M6/M3 metering standard does not affect the relative audible balance of sounds panned to one side versus the centre – that is determined solely by the panning law of the mixing console’s pan-pot.
  
  The M6 standard is deemed a simpler form of metering for untrained broadcasters to use as it keeps the M meter at '4' for Alignment Level and '6' for peaks, without the operator having to remember to subtract 3 dB.
   BULLET::::- Duplication and redirection requires setting up a VLAN (Virtual LAN) and including all the phones within the VLAN. Then SPAN that VLAN. This will allow recording all in/out bound traffic and internal traffic. The disadvantage is that not all phones at times are on a VLAN or the same VLAN, so multiple SPANS are needed. Another method is to use a concept called RSPAN (remote SPAN), in which the VLAN's that are set to SPAN are trunked across switches to a receiving switch.
  
  Section::::Challenges.
  