ING Telecom - VOIP provider

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Ing Telecom finishes great quantities of traffic of telecommunications in the whole world. Thanks to our wholesale volume, we have a certain weight in the traffic minutes wholesale, we obtain better tariffs on the international destinations and can thus make some profit our customers.
Ing Telecom envisages of A to Z the termination voice by interconnections with suppliers of level 1. The quality of the interconnections is quite simply the best possible one. ASR ASR: rate of success by calls & ACD ACD: average time of the calls, PPD PPD: time between the launching of the call and the first ringing runs are the witnesses of the quality of our system. Ing Telecom profits from a support network 24 / 7 - a center of operation ensures the constant quality of our service.

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• Protocoles: SIP SIP: Open standard of widest VoIPinteropérable aiming at becoming the multi-media standard of telecommunications (its,
  image, etc). Native of world Internet (HTTP). Session Initiation Protocol (of which the initials is SIP) is a protocol
  standardized and standardized by the IETF (described by the RFC 3261 which makes obsolete the RFC 2543) which was
  designed to establish, modify and finish sessions multi-media. It takes care of the authentification and the localization
  of the multiple participants. It also takes care of the negotiation on the types of media usable by the various participants
  by encapsulating messages SDP (Session Protocol Description). SIP does not transport the data exchanged during
  the session like the video voice or it. SIP being independent of the transmission of the data, any type of data and
  protocols can be used for this exchange. However protocol RTP(Real-time Protocol Transport) generally ensures the audio
  and video sessions. SIP replaces H323. SIP gradually is the open standard of VoIP (Voice Over IP, voice on IP) interopérable
  widest and aims at becoming the multi-media standard of telecommunications (its, image, etc). SIP is not only intended
  for VoIP but for many other applications such as the visiophonie, the instantaneous transport, virtual reality or even the
  video games & H323 H323 is a whole of
  standards UIT-T which define the protocols making it possible to
  establish an audio or video communication on a data-processing
  network. H323 is dated an enough protocol which is currently exceeded
  by the SIP - Session Protocol Initiation. One of the advantages of
  the SIP is its simplicity and its resemblance with protocols HTTP and
  smtp This is why the majority of the VoIP material available today
  meets standards SIP. The older equipment on the other hand will
  follow the standards of the protocol H323. & IAX IAX is a protocol of
  voice on IP resulting from the project of open PABX Asterisk source.
  IAX allows the communication between customer and waiter like between
  waiters. IAX is sometimes considered to be more powerful than SIP
  because it was conceived for the control and the multi-media
  transmission of flow with a lower flow (in particular for the voice)
  and integration in the NATés networks. Indeed IAX uses one port UDP:
  4569, for indication and the data. Name IAX is often used to speak
  about version 2 of the protocol. Indeed, the first version is not
  used practically any more. Its weaknesses are its youth and its
  not-standardization, although it is used more and more. IAX2 meets
  more and more success, which testifies the greatest availability of
  compatible material. IAX2 does not constitute however a real threat
  for SIP, because this last is already well installed in industry.
  IAX2 positions on the other hand like an increasingly credible
  alternative.
• Codecs Codec : Portemanteau word codec comes from '
  coder-decoder' and indicates a process able to compress or of
  décompresser a signal, analogical or numerical, in a format of data.
  This process can exist in material or software form. For example,
  in the middle of the twentieth century, word codec was used to
  indicate a type of material making it possible to code an analogical
  signal in PCM and to decode it in return. The codecs encodent flows
  or signals for the transmission, the storage or the encoding of data.
  Of another dimensioned, they decode these flows or signals for
  edition or visionnage. The goal first of the codecs is to be able to
  treat a maximum of data with a minimum of resources. They are used
  for applications like telephony or the vidéos-conference supported: G711 G711 is a standard of audio compression of the UIT-T, based on the µ-law and A-law.
  • sampling: 300 to 3400 Hz (band-width of the telephone)
  • Band-width on the network: 64 or 56 kbit/s
  • Standard of
    coding: MIC (Pulse modulation and Coding)
  Especially used for the
  videoconference in H.323 and H.320. Its principle rests on a nonlinear
  grid of quantification, making it possible to decrease the
  signal-on-noise ratio of the error of quantification for the sounds of
  low amplitude. A quantification on 8 bits in G.711 corresponds to a
  quantification on 12 bits in PCM with regard to the error of
  quantification. Note: The standard of audio compression G.711 is
  exceeded today, in term of band-width on the network, with the favour
  of others codecs of audio compression. It is present, today, only for
  reasons of compatibility and of interworking / G729 G729 is a less
  consumer in band-width than G.711. It is used to obtain a telephony
  of quality. The codec G.729 is: • used for the coding of the audio
  part of a videoconference, • also met to transport voice on IP on the
  WAN, • Used preferentially by the operators of telephony such as
  Annatel or others / G726 G726 is a standard of audio compression of
  the UIT-T. Pulse modulation and adaptive differential coding (MICDA)
  with 40, 32, 24, 16 kbit/s. / GSM / iLBC iLBC means: Internet Low Bitrate codec, low
  codec flow on Internet. The flow at exit of the codec is 13.33 kbps
  (screen every 30 ms) or 15,2 Kbps (screen all the 20 ms)ce which gives
  approximately 39 Kbps (at 20 ms), without suppression of silences, for
  the resulting flow on an Ethernet bond. (20ms & 30ms)
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