VoIP gateway The term sometimes also often used to refer to other functional elements, in this case is often called special VoIP gateways, as are positioned between IP networks to develop certain mapping capabilities, for example in the IP layer. Specific entities such as proxies VoIP, VoIP transcoders, translators VoIP network address, etc.., Fall into this category of VoIP gateways.
* Interconnection gateways in this context are essentially logical devices, although they may be, and indeed are physical devices, as discussed later. They have a number of attributes that characterize the volume and types of services they can provide, for example:
• Capacity, expresses the volume of service we can provide the gateway, being directly related to the number of ports (equal to the maximum number of simultaneous calls) and speed of access link.
• Signaling protocols supported, both on VoIP networks as on SNA networks.
• voice codecs used.
• encryption algorithms it supports.
• address range, which is the range or range of phone numbers through it is being accessed in the GSTN from the IP network. In relation to charging, this address range may or may not be split.
* In general, interconnection gateways must provide the following "mechanisms" or functions
• Adaptation of signage, basically has to do with the functions of establishing and terminating calls,
• Control of the media, is related to the identification, processing and interpreting service-related events generated by users or terminals
• Adaptation of means, according to network requirements.
The interconnection gateway or gateway function also develops media control, which deals with "manage" all the control information generated by the terminal. In the case of voice communications, information from user level control further highlight are the multifrequency tones (DTMF) which produces a conventional phone keypad (eg to interact with a voice server). However, given the characteristics of these signals, in the sense that they are in the audible range but are not voice signals, but tones it is necessary to pay particular attention to its transfer by hybrid connection that represents the gateway interconnection. The voice compression techniques introduce considerable speed low distortion in the DTMF, causing the reception and decoding for error in the receivers. Then, this requires that the audio signals and DTMF tones are separated in the gateway (if you have not already been in the issuer) and conducted independently by the receiver.
* There are two possible solutions for the transport of DTMF tones:
Transport "in-band" is to carry these tones, digitized and packaged with the protocols RTP / UDP, using a dedicated payload format.
* Transport "out of band" involves using a control channel means insurance (not UDP, but TCP) for transport of signals TDMF.
The transport of DTMF tones "in-band" is affected by the absence of guarantee delivery of packets on UDP offers, with dire consequences for the operation of the service in case of loss of a packet associated with a tone TDMF . An advantage is that the tones stay synchronized in time with respect to the voice.In contrast, the transport "out of band" although security gains in respect of the safe delivery of packages, they lose their specific signals in time relative to the voice stream. This is precisely the solution adopted in Recommendation H.323 through H.245 channel.