Dss «Kvant-e». Subscriber access subsystem. Algorithm of incoming call in slu-128.
Dss «Kvant-e». Signalling subsystem. Innersystem signalling, issc packet structure.
The term of subscriber signaling refers to signaling communicated between subscriber terminal and a telephone exchange.
For example, it includes the following signals:
direct current signals;
DTMF or DCLP
Intraexchange signaling carries control and address information between controllers. The information is transferred via the 16-th TC in the packets containing 16 bytes. It takes one multi-frame to transmit one packet.
TC
TC
DTU
CP
CD
SCM
TC
CS-8a
SSU
Trunk
SCU
TSS
SM
Figure 18 – Signaling path
Structure of ISSC packet is given below.
Figure 19 – Structure of ISSC packet
0 byte carries synchronization;
1 αβγ identifies packet’s length (usually equals 6); the S flag refers to changes in packet’s content compared to a previous one. K is a control bit.
abcd digits encode a dialed digit while 12345 digits encode a number of reserved TC and trunk.
The last two bytes (14 and 15) carry information about channel’s state and restart information if necessary.
Architecture of dss si-2000. Parameters and short characteristics of modules mlc and mca
MLC (Line Module version C) represents a hardware implementation of an Access Node.
According to configuration of AN and used software it is possible to implement the following types of Access Nodes.
1) Narrowband Access Node AN-NB (mini AN-NB)
It supports:
320 or 704 analogue subscriber lines;
160 to 320 digital subscriber lines with BRA;
10 or 20 digital subscriber lines with PRA;
16 or 32 El trunks to access other telephone exchanges of a network as well as SN or SAN;
it converts signaling.
AN-NB uses 3 or 4 El trunks to access SN or SAN with the help of the V5.2 protocol.
MCA (Central Module version A) is a hardware implementation of Switch Node. Its functions are listed below.
it creates connections between any time channels of any trunks without internal congestions;
it distributes user data according to address information;
call set up handling and disconnection;
synchronization of system's modules;
it accomplish interaction with Management Node.
Capacity of a switching network (for one MCA module) can be 240x240 El trunks.
The switching network of the Ts type is backed up, single-cascaded. All the access nodes (local and remote ones) use El trunks (and V5.2 protocol) to connect MCA. Quantity of trunks between an access node and MCA depends on capacity of the AN and averaged load on a subscriber line.
Architecture of si-2000 dss
Architecture of SI-2000 DSS is based on host equipment. It implements functions of centralized maintenance of all the distributed equipment facilities of the DSS including different switching modules and remote subscriber modules.
Network built upon SI-2000 equipment includes the following nodes:
Switch node (SN);
Access Node (AN);
Switch and access node (SAN) - RSwM;
Management Node (MN);
Switch Node
It is a set of group switching because it routes groups of channels. It controls all the access nodes, performs alarm monitoring and collect statistic and billing information.
MCA (Central Module version A) is a hardware implementation of Switch Node. Its functions are listed below.
it creates connections between any time channels of any trunks without internal congestions;
it distributes user data according to address information;
call set up handling and disconnection;
synchronization of system's modules;
it accomplish interaction with Management Node.
Capacity of a switching network (for one MCA module) can be 240x240 El trunks.
The switching network of the Ts type is backed up, single-cascaded. All the access nodes (local and remote ones) use El trunks (and V5.2 protocol) to connect MCA. Quantity of trunks between an access node and MCA depends on capacity of the AN and averaged load on a subscriber line.
Access Nodes
Access nodes provide interfaces for subscriber lines, perform BORSCHT functions (for analogue subscriber lines) provides PRA and BRA access for digital subscriber lines. Access nodes support subscriber signaling and concentrate subscriber traffic.
MLC (Line Module version C) represents a hardware implementation of an Access Node.
According to configuration of AN and used software it is possible to implement the following types of Access Nodes.
1) Narrowband Access Node AN-NB (mini AN-NB)
It supports:
320 or 704 analogue subscriber lines;
160 to 320 digital subscriber lines with BRA;
10 or 20 digital subscriber lines with PRA;
16 or 32 El trunks to access other telephone exchanges of a network as well as SN or SAN;
it converts signaling.
AN-NB uses 3 or 4 El trunks to access SN or SAN with the help of the V5.2 protocol.
Next Generation Networks. Definition of the Next Generation Network. Services in multiservice networks and quality of service. Nomenclature and characteristics of NGN services and their implementation (Triple Play Service).
The term of NGN network refers to a telecommunication network providing Triple Play Services (voice, data, and video). Such a network has flexible control features. It enables personalizing and creation of new services with the help of common network solutions based on universal transport network with distributed switching and QoS conception.
The term of multiservices network refers to a telecommunication network built according to the NGN conception providing Triple Play Services (voice, data, and video).
NGN services can be divided into the three following groups.
voice transfer services;
video transmission services (like images, telefax messages, color fax messages, video calls, video conferences, television services)
data transmission services (data exchange services, search for data and others).
A Next generation network (NGN) is a packet-based network able to provide services including Telecommunication Services and able to make use of multiple broadband, Quality of Service-enabled transport technologies in which service- related functions are independent from underlying transport-related technologies. It offers unrestricted access by users to different service providers.
Quality of service is the ability to provide different priority to different applications, users, or data flows, or to guarantee a certain level of performance to a data flow. For example, a required bit rate, delay, jitter, packet dropping probability and/or bit error rate may be guaranteed. Quality of service guarantees are important if the network capacity is insufficient, especially for real-time streaming multimedia applications such as voice over IP, online games and IP-TV, since these often require fixed bit rate and are delay sensitive, and in networks where the capacity is a limited resource, for example in cellular data communication.
In the field of telephony, quality of service was defined in the ITU standard X.902 as "A set of quality requirements on the collective behavior of one or more objects". Quality of Service comprises requirements on all the aspects of a connection, such as service response time, loss, signal-to-noise ratio, cross-talk, echo, interrupts, frequency response, loudness levels, and so on. A subset of telephony QoS is Grade of Service (GOS) requirements, which comprises aspects of a connection relating to capacity and coverage of a network, for example guaranteed maximum blocking probability and outage probability.
QoS problems
Many things can happen to packets as they travel from origin to destination, resulting in the following problems as seen from the point of view of the sender and receiver:
Dropped packets
The routers might fail to deliver (drop) some packets if they arrive when their buffers are already full. Some, none, or all of the packets might be dropped, depending on the state of the network, and it is impossible to determine what will happen in advance. The receiving application may ask for this information to be retransmitted, possibly causing severe delays in the overall transmission.
Delay
It might take a long time for a packet to reach its destination, because it gets held up in long queues, or takes a less direct route to avoid congestion. In some cases, excessive delay can render an application such as VoIP or online gaming unusable.
Jitter
Packets from the source will reach the destination with different delays. A packet's delay varies with its position in the queues of the routers along the path between source and destination and this position can vary unpredictably. This variation in delay is known as jitter and can seriously affect the quality of streaming audio and/or video.
Out-of-order delivery
When a collection of related packets is routed through the Internet, different packets may take different routes, each resulting in a different delay. The result is that the packets arrive in a different order than they were sent. This problem requires special additional protocols responsible for rearranging out-of-order packets to an isochronous state once they reach their destination. This is especially important for video and VoIP streams where quality is dramatically affected by both latency and lack of isochronicity.
Error
Sometimes packets are misdirected, or combined together, or corrupted, while en route. The receiver has to detect this and, just as if the packet was dropped, ask the sender to repeat itself.
How does it work?
QoS works by slowing unimportant packets down, or in the cases of extreme network traffic, throwing them away entirely. This leaves room for important packets to reach their destination as quickly as possible. Basically, once your router is aware of how much data it can enqueue on the modem at any given time, it can "shape" traffic by delaying unimportant packets and "filling the pipe" with important packets FIRST, then using any leftover space to fill the pipe up in descending order of importance.
Since QoS cannot possibly speed up a packet, basically what it does is take your total available upstream bandwidth, calculate how much of the highest priority data it has, put that in the buffer, then go down the line in priority until it runs out of data to send or the buffer fills up. Any excess data is held back or "requeued" at the front of the line, where it will be evaluated in the next pass.
QoS priority levels
Priority level |
Traffic type |
0 (lowest) |
Best effort |
1 |
Background |
2 |
Standard (spare) |
3 |
Excellent Load (Business Critical) |
4 |
Controlled Load (Steaming Multimedia) |
5 |
Voice and Video (interface Media and Voice) [less than 100ms latency and jitter] |
6 |
Layer 3 Network Control Reserved Traffic [Less than 10ms latency and jitter] |
7 |
Layer 2 Network Control Reserved Traffic [Lowest latency and jitter] |
Triple play
In telecommunications, the triple play service is a marketing term for the provisioning of two bandwidth-intensive services, high-speed Internet access and television, and a less bandwidth-demanding (but more latency-sensitive) service, telephone, over a single broadband connection. Triple play focuses on a combined business model rather than solving technical issues or a common standard. However, single standards like G.hn do exist to deliver all these services on a common platform.
A so-called quadruple play service integrates mobility as well, often by supporting dual-mode GSM plus Wi-Fi cell phones that shift from GSM to Wi-Fi when they come in range of a home wired for the triple play service. Typical services of this kind, such as Rogers Home Calling Zone, allow the caller to enter and leave the range of their home Wi-Fi network with only the time they spend outside the range paid for on the GSM plan - the rest of it being routed over the IP network and paid at a flat rate per month. No interruption or need to authorize the shift is ever required - call setup takes place automatically as many times as the caller enters or leaves the range.
Figure 1 NGN application for Triple Play Solution
Functional architecture of NGN. Architecture of NGN and assignment of its components (AG, MG, SG, Softswitch). The generalized structure of the network based on Softswitch. Softswitches 4th and 5th classes. Signaling protocols in the NGN.
A Next generation network (NGN) is a packet-based network able to provide services including Telecommunication Services and able to make use of multiple broadband, Quality of Service-enabled transport technologies in which service- related functions are independent from underlying transport-related technologies. It offers unrestricted access by users to different service providers.
Next Generation Networks are based on Internet technologies including Internet Protocol (IP) and Multiprotocol Label Switching (MPLS). At the application level, Session Initiation Protocol (SIP) seems to be taking over from ITU-T H.323.
For voice applications, one of the most important devices in NGN is a Softswitch - a programmable device that controls Voice over IP (VoIP) calls. It enables correct integration of different protocols within NGN. The most important function of the Softswitch is creating the interface to the existing telephone network, PSTN, through Signaling Gateways and Media Gateways.
Signaling gateway (SG)
The SG provides the signaling interface between the IP network and the PSTN signaling network. It terminates SS7 links and provides Message Transport Part (MTP) Level 1 and Level 2 functionality. Each SG communicates with its associated circuit switch (CS) to support the end-to-end signaling for calls.
Access Gateway (AGW): It acts as the line side interface to the core IP network and connects subscribers with analog subscriber access, integrated services digital network (ISDN) subscriber access, V5 subscriber access, PABX and x digital subscriber line (xDSL) access.
Trunk Media gateway (IMG)
It resides between the circuit switched (CS) network and the IP network. It converts format between pulse code modulation (PCM) signal flow and IP media flow. It supports functions such as packet station, echo control etc. It can have integrated signaling gateway functionality also. The MGW can connect with devices, such as the PSTN exchange, private branch exchange (PBX), access network devices and base station controller (BSC).
The softswitch, also known as Media Gateway Controllers (MGC), Call Servers (CS) and Call Agents is the core device in the NGN. The Softswitch is located in the service provider’s network and handles call control and signaling functions, typically maintaining call state for every call in the network. A Softswitch interacts with Application Servers to provide services that are not directly hosted on Softswitch. Important functions of Softswitch are:
Call control
Media gateway access control
Signalling Gateway Control
Border Gateways control
Resource allocation
Protocol processing
Routing
-Authentication
Charging
Softswitches also act as Signaling Switching Point (SSP) to provide access to IN services to SIP users.
The NGN architecture incorporates:
Support for multiple access technologies: The NGN architecture offers the configuration flexibility needed to support multiple access technologies.
Distributed control: This will enable adaptation to the distributed processing nature of packet-based networks and support location transparency for distributed computing.
Open control: The NGN control interface is open to support service creation, service updating, and incorporation of service logic provision by third parties.
Independent service provisioning: The service provisioning process is separated from transport network operation by using the above-mentioned distributed, open control mechanism. This is intended to promote a competitive environment for NGN development in order to speed up the provision of diversified NGN services.
Support for services in a converged network: This is needed to generate flexible, easy-to-use multimedia services, by tapping the technical potential of the converged, fixed mobile functional architecture of the NGN.
Enhanced security and protection: This is the basic principle of an open architecture. It is imperative to protect the network infrastructure by providing mechanisms for security and survivability in the relevant layers.
The figure 5 shows the control and media streams in NGN environment. The media streams consist of audio, video or data, or a combination of any of them. Media stream conveys user or application data (i.e., a payload) but not control data. It is transported through RTP/RTCP. Control signaling messages are transported by control streams using signaling protocols like SIGTRAN, H.248, H.323, SIP etc.
H.323
H.323 is an ITU Recommendation that defines "packet-based multimedia communications systems." It defines a distributed architecture for creating multimedia applications, including VoIP. The H.323 protocol is best known as the original call signaling protocol that made real time voice and video over IP possible. Being the first solution to work, H.323 is the most widely deployed protocol in the market and through its veteran status and wide acceptance provides telecommunication equipment with the benefits of a highly mature and completely interoperable signaling solution.
SIP
Designed by the IETF, the Session Initiation Protocol (SIP) is an application-layer control protocol for multimedia communication over IP network. It is used for creating, modifying and terminating two party sessions, multiparty sessions and multicast sessions (one sender and many receivers). These sessions include audio, video and data for multimedia conferences, instant messaging, Internet telephone calls, distance learning, telemedicine, multiparty real time games etc.
Sip defines telephone numbers as URLs (Uniform Resource Locators), so that web pages can contain them. This allows a click on a link to initiate a telephone call. These addresses take the form of user@host, similar to e-mail addresses. The user part, which is left of the sign, may be user name or a telephone number and host part, which is right of the sign, is a domain name or IP address. SIP addresses may be obtained out-of-band, learned via media gateways, recorded during earlier conversations, or guessed (since they’re often similar to E-mail addresses. SIP may be used in conjunction with other call setup & signaling protocols and has a verity of other features like caller reachability, call screening, encryption and authentication etc.
MGCP
Media Gateway Control Protocol (MGCP) is a control protocol that uses text or binary format messages to setup, manage, and terminate multimedia communication sessions in a centralized communications system. This differs from other multimedia control protocol systems (such as H.323 or SIP) that allow the end points in the network to control the communication session. MGCP is specified in RFC 2705. MGCP is, in essence, a master/slave protocol, where the MGs are expected to execute commands sent by the MGCs.
H.248
H.248 is an ITU Recommendation that defines ‘Media Gateway Control Protocol’. It is the result of a joint collaboration between the ITU and the IETF. It is also referred to as IETF RFC 2885 (MEGACO), which defines a centralized architecture for creating multimedia applications, including VoIP. In many ways, H.248 builds on and extends MGCP. It is used as a media gateway control protocol between a Media Gateway Controller (MGC) and a Media Gateway (MG).
SIGTRAN
SIGTRAN (Signaling Transport) is a protocol stack defined by the SIGTRAN workgroup of the Internet Engineering Task Force (IETF) for transport of switched circuit network (SCN) signaling over IP networks.
This protocol stack supports the inter-layer standard primitive interface defined in SCN signaling protocol hierarchy model to ensure utilization of the existing SCN signaling application without modification. It uses the standard IP transport protocol as the transmission bottom layer, and satisfies the special transmission requirements of SCN signaling by adding its own functions.
PARLAY/ JAIN
Parlay/JAIN is a suite of open, standard, APIs designed to facilitate easier access to core network capabilities from outside of the network. The opening up of the network in a secure manner by such APIs allows the existence of new business models, which allow applications to be developed and provided by vendors outside of the network operator’s domain.
