Telephony through a (sometimes called ADSL line Voice over xDSL) can be done in several ways, even if today,.
ADSL providers seems all adopt the same technology to set up their telephone services.
Telephone conversations are possible without the addition of specific equipment at the Subscriber, and are directly transmitted by the classic RTC (via the network to a phone operator).
Provide telephone service of this type requires access to the frequencies of the telephone line.
This is possible only if the supplier has made a full unbundling of the Subscriber's telephone line, or if the supplier is increasingly an alternative phone operator.
Voice over IP is a mechanism that allows to pass audio conversations between several devices interconnected between them via a computer network supporting the IP protocol.
Worldwide DSL, you can also find it under the acronym VoMSDN (Voice over MultiService Data Networks).
It basically consists of three modules :
• a module of digitization of voice, which is to try to convert the analog voice into digital data (and), possibly using compression algorithms.
• a protocol of control of the conversation, which allows to establish a conversation between several participating, and who is responsible for negotiating the parameters of configuration of this conversation.
• an audio data transport protocol, which encapsulates the voice scanned in IP packets to carry it across the network, possibly using a network telephone Commute 10 : network of fixed telephone, in which a subscriber station is connected to a telephone exchange by a pair of wires powered Central battery. Central offices are themselves interconnected by links providing a flow of 2 Mbps or more.
Finally, the electronic circuits of digitization of the voice being today expensive (because it's a technology damped, and they are now mass produced).
It allows to include directly in the STB such circuits at lower cost.
Thus, it is this type of technique that is almost exclusively used by suppliers to offer their phone service.
The VoIP is only a concept, and there are now several incompatible implementations between them.
However, they work globally with the same types of network elements :
• terminals : this can be either a computer with an IP connection or an IP phone, or a device to scan the signals of a classic phone (the case of a STB for example). It is through them that users initiate and receive phone conversations.
• registers : they allow you to associate a terminal (via its IP address) to a ID more simple to use and known by users who want to contact this terminal. This association can be done using a more or less complex identification mechanism.
• the proxy : these are elements which allow the linking of two terminals which do not (i.e., the calling terminal does not know the IP address of the terminal to call). They are generally related to records (or combined), and can control the access to a terminal on the basis of a rights management policy.
• gateways : they ensure the inter-connecting between a specific VoIP network and another type of telephone network (PSTN, VoIP, VoATM, etc.).
• multipoint control units)Multipoint Control Unit) : they allow to manage the communications involving more than two participants.
These implementations can usually perform three types of conversations :
• point to point : terminals that want to communicate (to a maximum of two terminals by conversation) connect directly between them, by performing a phase denegociation to determine the parameters of the communication.
Once you have initialized the communication, audio (only in the case of VoIP) can flow between two terminals, using an isochronous data transfer protocol.
This type of communication requires the IP address of the terminal that a user wants to join, which is not always possible (especially when one of the two participants is located behind a router NAT).
• point to point, via a proxy : here, all terminals register with the registry in order to be associated with an identifier known to users likely to call.
When a terminal needs to contact another, it then uses the proxy giving him the identifier of the terminal it wants to contact.
The proxy then uses the registry to determine the IP address of the terminals to be attached, and determines if this terminal is free and accessible by the calling terminal.
If so, the communication between the two terminals is point to point.
• Multipoint : in a communication between two terminals, an MCU is used to manage the networking of the various participants.
It allows to specify the maximum number of participants, the flow of communication, communication, ID etc...
The current implementations of VoIP are designed, not not to make conversations audio only, but to manage and carry multimedia communications, which may include as many votes as the video or data (text, drawings, etc.).
• H.323 : this standard is a set of communication protocols of voice, image and data over IP. It is a protocol which was developed by ITU - T in 96. It is derived from the H.320 protocol used on ISDN. It is based on the operation of the phone classic (at the time), and is indeed quite complex and very rigid. It is more and more replaced by the Protocol SIP, simple to use and much more modular.
• SIP (Session Initiation Protocol) : this is a protocol standard and standardized by the IETF (RFC 3261) which was designed to establish, modify and terminate multimedia sessions. He is responsible for authentication and location of multiple participants. He is also responsible for the negotiation on the types of media that can be used by different participants. SIP does not carry the data exchanged during the session as the voice or video. SIP is independent of the transmission of the data, any type of data and protocols can be used for this Exchange.
VoIP telephony services generally use the Protocol SIPin association with other signage protocols that allow to simply adapt traditional telephony signals (from the PSTN or a classic phone) VoIP controls signals. So, we can for example find the SIP - T protocols (Session Initiation Protocol for Telephones) and MGCP (Media Gateway Control Protocol) in the Free ISP VoIP device.
The Protocol SIP-T allows to adapt the signals (or destination) of the RTC to use VoIP, and MGCP allows to adapt the signals (or destination) a traditional telephone so that the STB can use VoIP, without making a big conversion job.
Audio data traveling by the RTP Protocol, between a set-top box and a switch in the case of an internetwork communication, or directly between two Box
the switch that makes the bridge with the second participant of the communication network.
VoIP is a technique particularly suited for telephony in the Triple Play offers, share its low cost (due mainly to the fact that it re - use the ISP network) and ease of integration into the STB.
The VoATM (also known as Voice traffic over ATM) is a technique that allows you to take advantage of the features of multi-flow of ATM transport.
The DSLAM is most of the time connected by ATM connections to the ISP, it may seem more sensible to use the VoATM to pass the audio stream ADSL rather than use the VoIP protocol.
In addition, ATM allows booking of throughput, that does not have IP, this could be a big problem in type isochronous transfers.
However, the VoATM is not used by the ADSL providers that offer phone service, probably for reasons of improbable developments. That is why this technology is cited here only at informative title, and will not be developed more.
Finally, it should be noted that this technology is however used by some American providers.