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2009 August | Alice in Telecomland

VoIP

Posted by Arifa Anees | Posted in telecom | Posted on 15-08-2009

Voice over Internet Protocol is a transmission technology for delivery of voice communications over the Internet or other packet-switched networks frequently encountered terms with VoIP are IP telephony and Internet telephony, as well as voice over broadband, broadband telephony, and broadband phone. VoIP systems usually interface with the traditional public switched telephone network (PSTN) to allow for transparent phone communications worldwide. VoIP can be a benefit for reducing communication and infrastructure costs by routing phone calls over existing data networks and avoiding duplicate network systems. Voice-over-IP systems carry telephony speech as digital audio, typically reduced in data rate using speech data compression techniques, packetized in small units of typically tens of milliseconds of speech, and encapsulated in a packet stream over IP.

VoIP components perform the same features as the PSTN network, which are

  • Signaling
  • Database services
  • Call connect and disconnect (bearer control)
  • CODEC operations

 

·         SIGNALING

The signaling in a VoIP network activates and coordinates the various components to complete a call. Although the underling nature of the signaling is the same, there are some technical and architectural differences in a VoIP network.

Signaling in a VoIP network is accomplished by the exchange of IP datagram messages between the components. The format of these messages is covered by any number of standard protocols. Regardless of which protocol and product suites that is used, these message streams are critical to the function of a voice-enabled network and might need special treatment to guarantee their delivery.

·         DATABASE SERVICES

Database services are a way to locate an endpoint and translate the addressing that two networks use. For example, the PSTN uses phone numbers to identify endpoints, while a VoIP network could use an IP address (address abstraction could be accomplished with DNS) and port numbers to identify an endpoint. A call control database contains these mappings and translations. Another important feature is the generation of transaction reports for billing purposes. You can employ additional logic to provide network security, such as to deny a specific endpoint from making overseas calls on the PSTN side. This functionality, coupled with call state control, coordinates the activities of the elements in a VoIP network.

·         CALL CONNECT AND DISCONNECT (BEARER CONTROL)

The connection of a call is made by two endpoints opening communications sessions between each other. In the PSTN, the public or private switch connects logical DS-0 channels through the network to complete the calls. In a VoIP implementation, this connection is a multimedia stream (audio, video, or both) transported in real time. This connection is the bearer channel and represents the voice or video content being delivered. When communication is complete, the IP sessions are released and optionally network resources are freed.

·         CODEC OPERATIONS

Voice communication is analog, while data networking is digital. The process of converting analog waveforms to digital information is done with a coder-decoder (CODEC, which is also known as a voice coder-decoder [VOCODER]). There are many ways an analog voice signal can be transformed, all of which are governed by various standards. Most of the conversions are base on pulse coded modulation (PCM) or variations.

In addition to performing the analog to digital conversion, CODECs compress the data stream, and provide echo cancellation. The bandwidth savings for the voice services can come in several forms and work at different levels. For example, analog compression can be part of the encoding scheme (algorithm) and does not need further digital compression from the higher working layers of the media gateway application. Another way to save bandwidth is the use of silence suppression, which is the process of not sending voice packets between the gaps in human conversations. Using compression or silence suppression can result in sizable bandwidth savings. However, there are some applications that could be adversely affected by compression. Compression schemes can interfere with the functioning of modems by confusing the constellation encoding used. The result could be modems that never synchronize or modems that exhibit very poor throughput. Some gateways might implement some intelligence that can detect modem usage and disable compression.

How VoIP Works

VoIP services convert voice into a digital signal that travels over the Internet. If calling a regular phone number, the signal is converted to a regular telephone signal before it reaches the destination. VoIP can allow making a call directly from a computer, a special VoIP phone, or a traditional phone connected to a special adapter. In addition, wireless “hot spots” in locations such as airports, parks, and cafes allow you to connect to the Internet and may enable to use VoIP service wirelessly.

VoIP APPLICATIONS

VoIP could be applied to almost any voice communications requirement, ranging from a simple inter-office intercom to complex multi-point teleconferencing/shared screen environments, passing through applications like Net2Phone and voice mails. Some examples of VoIP applications are:

  • Remote access from a branch office: A small office could gain access to corporate voice, data, and facsimile services using the company’s Intranet. This may be useful for home-based agents working in a call centre. An example can be a bank that wants to reduce costs and combine traffic to provide voice and data access to the main office. This is accomplished by using a packet network to provide standard data transmission while at the same time enhancing it to carry voice traffic along with the data.
  • Internet-aware telephones: Ordinary telephones can be enhanced to serve as an Internet access device as well as providing normal telephony, using VoIP technology. The Internet is so rich in resources and the telephone is relatively poor, though that much popular and useful. Using VoIP, directory services, for example, could be accessed over the Internet by submitting a name and receiving a voice reply. The telephone can be used to query database for any information, including membership details to communication companies.
  • PSTN gateway: Interconnection of the Internet to the PSTN can be accomplished using a gateway either integrated into a PBX (the iPBX) or provided, for example, would have access to the public network by calling a gateway at a point close to the destination (thereby minimizing long distance charges).
  • Internet call centre access: Access to call centre facilities via the Internet is emerging as a valuable adjunct to electronic commerce applications. Internet call centre access would enable a customer who has questions about a product being offered over the Internet to access customer service agents online. VoIP can be further be used to interconnect different call centers, thereby coordinating the work between them.
  • Voice calls from a mobile PC via the Internet: Calls to the office can be achieved using a multimedia PC that is connected via the Internet. One example would be using the Internet to call from a hotel instead of using expensive hotel telephones. This could be ideal for submitting or retrieving voice messages. All you need is a PC (laptop), equipped with the necessary hardware and software to allow for VoIP application.

Voice over Internet Protocol (VoIP) is a technology that allows making voice calls using a broadband Internet connection instead of a regular (or analog) phone line. Some VoIP services may only allow calling other people using the same service, but others may allow calling anyone who has a telephone number - including local, long distance, mobile, and international numbers. Also, while some VoIP services only work over computer or a special VoIP phone, other services allow you to use a traditional phone connected to a VoIP adapter.

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MPLS

Posted by Arifa Anees | Posted in telecom | Posted on 09-08-2009

MPLS stands for Multiprotocol Label Switching. MPLS gives network operators a great deal of flexibility to divert and route traffic around link failures, congestion, and bottlenecks. Multi Protocol Label Switching (MPLS) is a data-carrying mechanism that belongs to the family of packet-switched networks. MPLS operates at an OSI Model layer that is generally considered to lie between traditional definitions of Layer 2 (Data Link Layer) and Layer 3 (Network Layer), and thus is often referred to as a “Layer 2.5″ protocol. It was designed to provide a unified data-carrying service for both circuit-based clients and packet-switching clients which provide a datagram service model. It can be used to carry many different kinds of traffic, including IP packets, as well as native ATM, SONET, and Ethernet frames.

A number of different technologies were previously deployed with essentially identical goals, such as frame relay and ATM. MPLS technologies have evolved with the strengths and weaknesses of ATM in mind. Many network engineers agree that ATM should be replaced with a protocol that requires less overhead, while providing connection-oriented services for variable-length frames. MPLS is currently replacing some of these technologies in the marketplace. It is highly possible that MPLS will completely replace these technologies in the future, thus aligning these technologies with current and future technology needs.

MPLS was originally proposed by a group of engineers from Ipsilon Networks, but their “IP Switching” technology, which was defined only to work over ATM, did not achieve market dominance. Cisco Systems, Inc. introduced a related proposal, not restricted to ATM transmission, called “Tag Switching”. It was a Cisco proprietary proposal, and was renamed “Label Switching”. It was handed over to the IETF for open standardization. The IETF work involved proposals from other vendors, and development of a consensus protocol that combined features from several vendors’ work. MPLS works by prefixing packets with an MPLS header, containing one or more ‘labels’. This is called a label stack. Each label stack entry contains four fields:

• 20-bit label value.
• 3-bit field for QoS (Quality of Service) priority
• 1-bit bottom of stack flag. If this is set, it signifies that current label is the last in the
stack
• 8-bit TTL (time to live) field.

These MPLS-labeled packets are switched after a Label Lookup/Switch instead of a lookup into the IP table. When MPLS was conceived, Label Lookup and Label Switching were faster than a RIB lookup because they could take place directly within the switched fabric and not the CPU.
The entry and exit points of an MPLS network are called Label Edge Routers (LER), which, respectively, push an MPLS label onto the incoming packet and pop it off the outgoing packet. Routers that perform routing based only on the label are called Label Switch Routers (LSR). Labels are distributed between LERs and LSRs using the “Label Distribution Protocol” (LDP). Label Switch Routers in an MPLS network regularly exchange label and reach ability information with each other using standardized procedures in order to build a complete picture of the network they can then use to forward packets. Label Switch Paths (LSPs) are established by the network operator for a variety of purposes, such as to create network-based IP Virtual Private Networks or to route traffic along specified paths through the network. In many respects, LSPs are no different than PVCs in ATM or Frame Relay networks, except that they are not dependent on a particular Layer 2 technology.

In the specific context of an MPLS-based Virtual Private Network (VPN), LSRs that function as ingress and/or egress routers to the VPN are often called PE (Provider Edge) routers. Devices that function only as transit routers are similarly called P (Provider) routers. The job of a P router is significantly easier than that of a PE router, so they can be less complex and may be more dependable because of this. When an unlabeled packet enters the ingress router and needs to be passed on to an MPLS tunnel, the router first determines the forwarding equivalence class (FEC) the packet should be in, and then inserts one or more labels in the packet’s newly-created MPLS header. The packet is then passed on to the next hop router for this tunnel. When a labeled packet is received by an MPLS router, the topmost label is examined. Based on the contents of the label a swap, push (impose) or pop (dispose) operation can be performed on the packet’s label stack. Routers can have prebuilt lookup tables that tell them which kind of operation to do based on the topmost label of the incoming packet so they can process the packet very quickly.

In a swap operation the label is swapped with a new label, and the packet is forwarded along the path associated with the new label.

In a push operation a new label is pushed on top of the existing label, effectively “encapsulating” the packet in another layer of MPLS. This allows hierarchical routing of MPLS packets. Notably, this is used by MPLS VPNs.

In a pop operation the label is removed from the packet, which may reveal an inner label below. This process is called “de-capsulation”. If the popped label was the last on the label stack, the packet “leaves” the MPLS tunnel. This is usually done by the egress router. During these operations, the contents of the packet below the MPLS Label stack are not examined. Indeed transit routers typically need only to examine the topmost label on the stack. The forwarding of the packet is done based on the contents of the labels, which allows “protocol-independent packet forwarding” that does not need to look at a protocol-dependent routing table and avoids the expensive IP longest prefix match at each hop.

At the egress router, when the last label has been popped, only the payload remains. This can be an IP packet, or any of a number of other kinds of payload packet. The egress router must therefore have routing information for the packet’s payload, since it must forward it without the help of label lookup tables. An MPLS transit router has no such requirement.

MPLS versus IP

MPLS cannot be compared to IP as a separate entity because it works in conjunction with IP and IP’s IGP routing protocols. MPLS gives IP networks simple traffic engineering, the ability to transport Layer 3 (IP) VPNs with overlapping address spaces, and support for Layer 2 pseudo wires. Routers with programmable CPUs and without LSP can either be

(a) Explicitly configured hop by hop,
(b) Dynamically routed by the Constrained Shortest Path First CSPF algorithm,
(c) Configured as a loose route that avoids a particular IP or that is partly explicit and partly dynamic.

In a pure IP network, the shortest path to a destination is chosen even when it becomes more congested. Meanwhile, in an IP network with MPLS Traffic Engineering CSPF routing, constraints bandwidth of the traversed links can also be considered, such that the shortest path with available bandwidth will be chosen.

MPLS versus Frame Relay

Frame relay aimed to make more efficient use of existing physical resources, which allow for the under provisioning of data services by telecommunications companies to their customers, as clients were unlikely to be utilizing a data service 100 percent of the time. Telcos often sell frame relay to businesses looking for a cheaper alternative to dedicated lines. Many customers are likely to migrate from frame relay to MPLS over IP or Ethernet within the next two years, which in many cases will reduce costs and improve manageability and performance of their wide area networks.

MPLS versus ATM

Both MPLS and ATM provide a connection-oriented service for transporting data across computer networks. In both technologies, connections are signaled between endpoints, connection state is maintained at each node in the path, and encapsulation techniques are used to carry data across the connection.

The most significant difference is in the transport and encapsulation methods. MPLS is able to work with variable length packets while ATM transports fixed-length (53 byte) cells. Packets must be segmented, transported and re-assembled over an ATM network. MPLS simply adds a label to the head of each packet and transmits it on the network.
An MPLS connection (LSP) is unidirectional. ATM point-to-point connections (Virtual Circuits), on the other hand, are bi-directional, allowing data to flow in both directions over the same path. Both ATM and MPLS support tunneling of connections inside connections.

The biggest single advantage that MPLS has over ATM is that it was designed from the start to be complementary to IP. Modern routers are able to support both MPLS and IP natively across a common interface allowing network operators great flexibility in network design and operation. ATM’s incompatibilities with IP require complex adaptation, making it comparatively less suitable for today’s predominantly IP networks.

MPLS provides networks with a more efficient way to manage applications and move information between locations. With the convergence of voice, video and data applications, business networks face increasing traffic demands. MPLS enables class of service (CoS) tagging and prioritization of network traffic, so administrators may specify which applications should move across the network ahead of others. This function makes an MPLS network especially important to firms that need to ensure the performance of low-latency applications such as VoIP and their other business-critical functions.

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