Wednesday, November 25, 2009

Voice over IP Gateway

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.

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soni2006 on HubPages

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Tuesday, November 24, 2009

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soni2006 on HubPages

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Monday, November 23, 2009

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soni2006 on HubPages

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What VoIP H.323 comprises of?

The VoIP/H.323 comprises a set of standards and is supported by a series of protocols covering the different aspects of communication:

* Address:

1. RAS (Registration, Admission and Status). Communications protocol that enables an H.323 station locate another station through the H.323 Gatekeeper.

2. DNS (Domain Name Service). Service name resolution to IP addresses for the same purpose that the RAS protocol but through a DNS server


1. Initial Q.931 Call Signaling

2. H.225 Call Control: signaling, registration and admissions, and packetization / synchronization of the stream (flow) voice

3. H.245 Control Protocol messages to specify opening and closing of channels for voice streams

* Voice Compression:

1. Required: G.711 and G.723

2. Optional: G.728, G.729 and G.722

* Transmission of Voice:

1. UDP. The transmission takes place on UDP packets, but UDP does not provide for data integrity, use of the bandwidth is higher than with TCP.

2. RTP (Real Time Protocol). Manages aspects of timing, scoring UDP packets with the information necessary for the efficient delivery of them at the reception.

* Transmission Control:

* RTCP (Real Time Control Protocol). It is mainly used to detect situations of network congestion and take, where appropriate, corrective action.

VoIP protocol stack

We can now from a number of items already available in the market and, according to different designs allow us to build VoIP applications. These elements are:

IP Phones.

PC Adapters.

Telephone Hubs.

Gateways (gateways PSTN / IP).


Multiple audio units. (MCU Voice)

Directory Services.

For more info read my hub:
Wireless Communication Protocols: GSM, H.323, PBX, PCM, WAN, LAN, Routers

VoIP Phones - How Voice over IP Works?

Actually, the integration of voice and data on one network is an old idea, as long solutions have emerged from different manufacturers, using multiplexers, allow the use of WAN data of companies (typically point connections to-point and frame-relay) for the transmission of voice traffic. The lack of standards and long repayment period of such solutions has ensured widespread deployment of them.

Undeniably, the definitive establishment of the IP protocol from the domestic business areas and the emergence of a standard, VoIP could not be expected. The emergence of VoIP along with the lowering of the DSP's (Digital Signal Processor), which are key in the compression and decompression of voice, are the elements that have made possible the launch of these technologies. For this boom there are other factors, such as the emergence of new applications or definitive commitment VoIP vendors like Cisco Systems and Nortel-Bay Networks. Moreover phone operators are offering or plan to offer in the near future, quality IP services to businesses.

It said so far, we see that we can find three types of IP networks:

* Internet. The current state of the network does not allow professional use for voice traffic.

* Public IP Network. The operators offer businesses the connectivity to interconnect their local area networks as IP traffic is concerned. It can be considered as similar to the Internet, but with a higher quality of service and significant security enhancements. There are operators even offer guarantees low delay and / or bandwidth, making them very interesting for voice traffic.

* Intranet. The IP network implemented by the company itself. Usually consist of several LAN (switched Ethernet, ATM, etc. ..) that are interconnected through WAN Frame-Relay/ATM type, point to point lines, ISDN remote access, etc.. In this case the company has control over virtually all network parameters, making it ideal for use in the transport of voice.

Due to the existing standard of ITU-T H.323, which covered most of the requirements for the integration of voice, it was decided that H.323 as the basis of VoIP. Thus, VoIP is seen as a clarification of H.323, so that in case of conflict, in order to avoid a gap between standards, it was decided that H.323 would have priority over VoIP. The VoIP has focused on ensuring interoperability between equipment from different manufacturers, fixing issues such as silence suppression, voice coding and routing, and establishing new elements to enable connectivity to the traditional telephone infrastructure. These elements are basically concerned with directory services and the transmission of signaling tone multifrequency (DTMF).

Also read my latest hub:

Why should I use a VoIP phone when I have a traditional phone available?