Saturday, August 20, 2011

IP Telephony Vs Public Switched Telephone Network

Looking back at the public switched telephone network, which until now has roughly an access network including the wiring from the subscriber's home to the local exchanges and the necessary equipment, a transmission system which includes superior power and communication links between them.

As indicated above all the resources to intervene in the development of a telephone conversation cannot be used by another call until the first does not finish.

In IP telephony, fundamental change occurs in the transmission system: Now this task is carried out by a network based on IP protocol, packet switching, including the Internet. As for the access network may be the same as in the previous case, physically speaking (local loop), but in terms of services is evident that the advantage is geared towards the ability to exchange data, sending images, graphs and videos, as he is talking to someone.

The necessary elements so that they can make voice calls through an IP network depends largely on which terminal is used at both ends of the conversation.

These can be IP or IP terminals. Among the first is the IP phone, a multimedia computer, an IP fax, among the latter is a conventional telephone, a conventional fax, the former are able to deliver to its output the telephone conversation in IP packet format, as well as being part own IP network, the latter not lie, so you need an intermediate device to do this before connecting to the IP network transport.

It should be noted that in the event that one or both ends of the telephone communication are IP terminal, it is important to know how they are connected to the Internet. If you permanently, we can establish communication at any time. If a non-permanent, for example, through an Internet service provider (ISP) via conventional modem (dial-up access), communication is only possible at the time the dial-up user is connected Internet.

Voip Bandwidth Consumption

Bandwidth Consumption in Voip: Achieving high quality carry voice over IP telephone in real time is not an easy task to achieve because such work requires management capabilities that allow network traffic control, real time protocols (TCP/IP are not) and bandwidths "engaged" during the time it takes the completion of the call.

However, day to day limitations on the services of IP-based voice, are being overcome by two factors: improvements in compression algorithms (which allow the optimal use of bandwidth) and the sophistication and development of existing routing protocols (capable of taking into account the delay for each of the possible paths that can take the package to determine the best route you can take, provide reserves bandwidth while tough talk and give preference to packet processing within the router, so that those of high priority are processed first).

VoIP Gateways: Addressing, Routing, RSVP, ATM

Voip Addressing: Taking again the example of an intranet with IP addressing, we could see that the voice interfaces appear as additional IP hosts, as extensions of existing numbering scheme or as new IP addresses.

The translation of the dialed digits to the host IP PBX are made by means of the numbering plan. The destination phone number or any part of this is linked to the destination IP address. When the number is received from the PBX router compared with those who have already been linked with an IP address and are listed in the routing table, if any match the call will be routed to the IP host to which this related, after the connection is established, a link to the intranet is transparent to the subscriber.

Voip Routing: One of the strengths of IP is the sophistication and development of their routing protocols. A modern routing protocol like EIGRP, is able to take into account the delay for each of the possible paths that can take the package and determine the best route you can follow. Advanced features such as the use of routing policies and use of access list (access lists), makes it possible to create highly secure routing schemes for voice traffic.

RSVP: Can be used by VoIP gateways, so as to ensure that traffic will go through the net for the best and shortest way, this may include segments of networks such as ATM or switched LANs. Some of the most important developments are IP routing, development of so-called tag switching and IP switching other techniques.

The sample tag into widespread switching IP routing, policies and capabilities of RSVP over ATM and other transport high. Another benefit of the tag switching is the traffic handling capacity, which is necessary for efficient use of network resources. The traffic management (traffic engineering) can be used to shift the burden of this in different sectors of the network based on different predictions depending on time of day.

Voice over IP - VoIP Compression and Signaling

Voice Compression: The compression algorithms used in gateways analyze a block of PCM samples delivered by the voice encoder (voice codec). These blocks have a variable length depending on the encoder, for example the basic size of a block of G.729 algorithm is 10 ms, while the basic size of a block of G.723.1 algorithm is 30ms.

The chain analog voice is digitalized in PCM frame, and so delivered the same compression algorithm at intervals of 10 ms.

VoIP signaling: Has 3 distinct areas: signaling the PBX to the router, signaling between the router and router to the PBX signaling. For example for a corporate intranet, this appears as the backbone to the PBX, which will give the signal to intranet users. Therefore the PBX forwards the digits to the router in the same way that the digits had been forwarded to a central telephone switch.

When the remote router receives the call requesting Q.931, this sends a signal to the PBX. After the PBX sends an acknowledgment, the router sends the dialed digits to the PBX, and process a call acknowledgment to the source router.

In a network architecture is not connection-oriented (like IP), responsibility for establishing communication and signaling is the end stations (end stations). To successfully provide voice services through a IP re, it is necessary to make improvements in signaling.

For example, an H.323 agent is added to the router to provide support for the transport of audio and signaling networks. The Q.931 protocol is used for the establishment and disconnection of the call between terminals agents or h.323. RTCP (RTP Control Protocol) is used to establish channels of audio. A reliable protocol connection oriented, TCP is used between end stations to carry the signaling channels.

RTP transport protocol in real time, which is supported on UDP, is used for the transport of audio stream in real time. RTP uses UDP as a transport mechanism because it has less delay than TCP, and because voice traffic today, whether they are data or signaling, tolerate lower levels of loss and lack the ease of transmission.