What type of protocol is tdm

It is often the case that not all the channels in a TDM circuit are simultaneously active "off-hook" , and activity status may be determined by observation of the TDM signaling channel. Moreover, even during active calls, about half the time is silence that can be identified using voice activity detection VAD.

Using the variable- rate AAL2 mode, we may dynamically allocate channels to be transported, thus conserving bandwidth. A packet may be constructed by inserting PDUs corresponding to all active channels, by appending PDUs ready at a certain time, or by any other means.

Hence, more than one PDU belonging to a single channel may appear in a packet. In such cases, these fields MAY be set to zero. Otherwise, the CCS channel should be treated as an ordinary channel. If the FCS is incorrect, the frame is discarded; otherwise, the frame is sent after initial or final flags and FCS have been discarded and zero removal has been performed.

We saw in Section 3 that defects can be indicated by setting flags in the control word. This insertion of defect reporting into the packet rather than in a separate stream mimics the behavior of native TDM OAM mechanisms that carry such indications as bit patterns embedded in the TDM stream.

The flags are designed to address the urgent messaging, i. In the "trail extended" OAM scenario, if there is a defect e. If a break occurs in the ultimate link, the IWF itself will detect the loss of signal. In either case, IWF1 having directly detected lack of validity of the TDM signal, or having been informed of an earlier problem, raises the local "L" defect flag in the control word of the packets it sends across the PSN.

Unlike forward defect indications that are generated by all network elements, reverse defect indications are only generated by trail termination functions. When the L flag is set there are four possibilities for treatment of payload content. Alternatively, with structure- aware transport of channelized TDM one SHOULD fill the payload with "trunk conditioning"; this involves placing a preconfigured "out of service" code in each individual channel the "out of service" code may differ between voice and data channels.

The third possibility is to suppress the payload altogether. When IWF2 receives a local defect indication without M field modification, it forwards or generates if the payload has been suppressed AIS or trunk conditioning towards ES2 the choice between AIS and conditioning being preconfigured. If the M field indicates that the TDM has been marked as potentially recoverable, then implementation specific algorithms not herein specified may optionally be utilized to minimize the impact of transient defects on the overall network performance.

The second case is when the defect is in TDM network 2. In the trail terminated scenario this RDI is restricted to network 2.

The final possibility is that of a unidirectional defect in the PSN. In all cases, if any of the above defects persist for a preconfigured period default value of 2. Since TDM PWs are inherently bidirectional, a persistent defect in either directional results in a bidirectional service failure.

In addition, if signaling is sent over a distinct PW as per Section 5. In the following subsections we review additional aspects essential to successful TDMoIP implementation. A jitter buffer is a block of memory into which the data from the PSN is written at its variable arrival rate, and data is read out and sent to the destination TDM equipment at a constant rate.

Use of a jitter buffer partially hides the fact that a PSN has been traversed rather than a conventional synchronous TDM network, except for the additional latency. Customary practice is to operate with the jitter buffer approximately half full, thus minimizing the probability of its overflow or underflow.

Hence, the additional delay equals half the jitter buffer size. In order to handle infrequent packet loss and misordering, a packet sequence integrity mechanism MUST be provided. An example sequence number processing algorithm is provided in Appendix A.

While the insertion of arbitrary filler data may be sufficient to maintain the TDM timing, for telephony traffic it may lead to audio gaps or artifacts that result in choppy, annoying or even unintelligible audio.

Alternatively one MAY replay the previously received packet. When computational resources are available, implementations SHOULD conceal the packet loss event by properly estimating missing sample values in such fashion as to minimize the perceptual error. Timing Recovery TDM networks are inherently synchronous; somewhere in the network there will always be at least one extremely accurate primary reference clock, with long-term accuracy of one part in 1E This node provides reference timing to secondary nodes with somewhat lower accuracy, and these in turn distribute timing information further.

This hierarchy of time synchronization is essential for the proper functioning of the network as a whole; for details see [ G ][G]. When emulating TDM on a PSN, extracting data from the jitter buffer at a constant rate overcomes much of the high frequency component of this randomness "jitter".

The rate at which we extract data from the jitter buffer is determined by the destination clock, and were this to be precisely matched to the source clock proper timing would be maintained. Unfortunately, the source clock information is not disseminated through a PSN, and the destination clock frequency will only nominally equal the source clock frequency, leading to low frequency "wander" timing inaccuracies. In broadest terms, there are four methods of overcoming this difficulty. In the first and second methods timing information is provided by some means independent of the PSN.

In a third method, a common clock is assumed available to both IWFs, and the relationship between the TDM source clock and this clock is encoded in the packet. In the final method adaptive clock recovery the timing must be deduced solely based on the packet arrival times.

Example scenarios are detailed in [ RFC ] and in [ Y ]. Adaptive clock recovery utilizes only observable characteristics of the packets arriving from the PSN, such as the precise time of arrival of the packet at the TDM-bound IWF, or the fill-level of the jitter buffer as a function of time.

Due to the packet delay variation in the PSN, filtering processes that combat the statistical nature of the observable characteristics must be employed.

For some applications, more stringent jitter and wander tolerances MAY be imposed. Unless appropriate precautions are taken, undiminished demand of bandwidth by TDMoIP can contribute to network congestion that may impact network control protocols. Note that: 1.

The results of the packet loss measurement may not be a reliable indication of presence or absence of severe congestion if the PSN provides enhanced delivery.

This specification does not define exact criteria for detecting severe congestion or specific methods for TDMoIP shutdown or subsequent re-start. However, the following considerations may be used as guidelines for implementing the shutdown mechanism: 1. TDMoIP does not provide mechanisms to ensure timely delivery or provide other quality-of-service guarantees; hence it is required that the lower-layer services do so. Switches and routers which the TDMoIP stream must traverse should be configured to respect these priorities.

Security Considerations TDMoIP does not enhance or detract from the security performance of the underlying PSN, rather it relies upon the PSN's mechanisms for encryption, integrity, and authentication whenever required. The level of security provided may be less than that of a native TDM service. These methods are beyond the scope of this document.

Random initialization of sequence numbers, in both the control word and the optional RTP header, makes known-plaintext attacks on encrypted TDMoIP more difficult. Encryption of PWs is beyond the scope of this document. When using adaptive clock recovery, the timing performance of the emulated TDM circuit depends on characteristics of the PSN, and thus may be inferior to that of a native TDM circuit. When packet loss events are properly concealed, the effect on telephony channels will be perceptually minimized.

However, the bit error rate will be degraded as compared to the native service. Data in inactive channels is not transported in AAL2 mode, and thus this data will differ from that of the native service.

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This is referred to as an OC-1 Optical Carrier 1. The number indicates the total of DS-3 channel equivalents in the payload. The routing of portions of the SONET payload to multiple points must be planned and built into a routing table.

These plans can be executed as part of a program to restore service in the event of an outage in a portion of the network. Some early adopters of SONET attempted to use this feature to provide for "time-of-day" routing changes. Often users were disappointed with the results.

However, it is possible to combine DS-3s into a single channel. An OC-3C concatenated is a group of DS-3s combined into a single payload to allow for the total use of the OC-3 as a single data stream. ATM is a data-link layer protocol that permits the integration of voice and data, and provides quality of service QoS capabilities. This standards-based transport medium is widely used for access to a wide-area WAN data communications networks. ATM nodes are sometimes called "Edge Devices".

These Edge Devices facilitate telecommunications systems to send data, video and voice at high speeds. ATM uses sophisticated network management features to allow carriers to guarantee quality of service. Sometimes referred to as cell relay, ATM uses short, fixed-length packets called cells for transport. Information is divided among these cells, transmitted and then re-assembled at their final destination. Carriers also offer " Frame Relay " service for general data requirements that can accept a variable packet or frame size.

Frame Relay systems use variable cell packets based on the amount of data to be transmitted. This allows for a more efficient use of a data communications network. ATM services are offered by most carriers. A number of DOTs are using this type of service — especially in metropolitan areas — to connect CCTV cameras using compressed video , traffic signal systems, and dynamic message signs to Traffic Operations Centers.

The stable packet size is well suited for video transmission. ATM is generally not used by telephone companies for toll grade voice, although its stable packet size was developed to meet requirements for voice service. ATM devices typically have ports that allow for easy connectivity of legacy systems and the newer communications systems. In a private or enterprise network, as shown in figure , ATM is effectively used for voice and video transport as well as data.

ATM has fixed-length "cells" of 53 bytes in length in contrast to Frame Relay and Ethernet's variable-length "frames. By catering to both forms of network traffic, ATM can be used to handle an end user's entire networking needs, removing the need for separate data and voice networks.

The performance, however, can also be compromised, and the network may not be as efficient as dedicated networks for each service. ATM systems usually require DS-1 circuits, but can be made to work in a lower speed environment. ATM does have a reputation for being difficult to interface to an existing network. However, competent network technicians can usually overcome most difficulties.

Frequency Division Multiplexing FDM is used when large groups of analog voice or video channels are required. The available frequency bandwidth on an individual communications link is simply divided into a number of sub-channels, each carrying a different communication session. A typical voice channel requires at least 3 kHz of bandwidth. If the basic communication link is capable of carrying 3 megahertz of bandwidth, approximately voice channels could be carried between two points.

Frequency Division Multiplexing was used to carry several low speed less than bits per second data channels between two points, but was abandoned in favor of TDM which has an ability to carry more data channels with more capacity over greater distances with fewer engineering problems.

This type of system was used by Freeway Management Systems to carry video over coaxial cable. However, most coaxial systems have been replaced by fiber optic systems. Fiber has a greater bandwidth capability than coaxial cable, or twisted pair. The FDM scheme allows for multiple broadband video channels to be carried over a single strand of fiber. A beam of light is divided into segments called lambdas. These lambdas are actually different colors of light. Light transmitted over a fiber is normally a group of frequencies that can be used to create a single communication channel, or multiple channels.

The frequency group can be broken into several sub groups. The LASER output of a multiplexer is "tuned" to a specific set of frequencies to form a single communication channel.

These channels are then transmitted with other frequency groups via a wave division multiplexer. Unlike FDM, the information sent via the frequency groups is digital. DWDM systems can carry as many as 64 channels at 2. Ethernet is a packet based network protocol, invented by the Xerox Corporation, in , to provide connectivity between many computers and one printer. Ethernet was designed to work over a coaxial cable that was daisy-chained shared among many devices.

The most current configurations use twisted pair with devices networked in a star configuration. Each device has a direct connection to an Ethernet hub, or router, or switch. This system then provides each user with a connection to a printer, file server, another user computer peer , or any other device on the network. Ethernet works by setting up a very broadband connection to allow packets of data to move at high speed through a network.

With carriers now turning up more and more SIP interconnections, the possibility of a call being SIP from end to end is increasing. With SIP comes additional capabilities.

As part of the information packet that is sent when a call is placed, the name of the user who is calling is passed. If the call is SIP from end to end, then the terminating carrier will receive this information and will display this information for CNAM.

If the user name is blank which would happen if it passed through some gateway along the way. Again, this information would have been stripped out in the gateway conversion , the terminating carrier would then choose to either display just city and state or dip the national CNAM database. Terminating carriers can also choose to use their own database and gather the data on their own and not dip the CNAM database. There are currently two providers of a national CNAM database.

VeriSign and Targus Neustar.