Saturday 30 December 2023

5G Network Funcntion :- UDM

The 5G UDM, or User Data Management, is a critical component in 5G (fifth-generation) wireless networks. It plays a key role in managing user-related information and authentication within the network. Here's a brief overview:

  1. Authentication and Authorization:

    • The UDM is responsible for authenticating and authorizing users on the 5G network. It verifies the identity of subscribers and ensures that they have the necessary permissions to access specific services.
  2. User Profile Management:

    • The UDM maintains user profiles, which include information such as subscription details, service profiles, and network access policies. This information is crucial for providing personalized services to subscribers.
  3. Subscription Management:

    • It handles the creation, modification, and deletion of user subscriptions. This includes managing different service plans, data plans, and other subscription-related information.
  4. Integration with Authentication Server:

    • The UDM interfaces with the Authentication Server (AUSF) to perform authentication functions. It ensures that only authorized users gain access to the network.
  5. Security Functions:

    • The UDM contributes to the overall security of the 5G network by implementing security protocols and encryption mechanisms to protect user data and communication.
  6. User Plane Functionality:

    • In some cases, the UDM may also be involved in user plane functionality, particularly in scenarios where the user plane and control plane functions are partially converged.
  7. Interworking with Other Network Elements:

    • The UDM interacts with other network elements, such as the Access and Mobility Management Function (AMF), to facilitate seamless mobility and connectivity for users as they move within the network.
  8. Support for Network Slicing:

    • With the advent of 5G, network slicing has become a crucial concept. UDM supports the creation and management of network slices, allowing for the customization of services and resources for different use cases.

In summary, the 5G UDM is a central element in 5G networks, managing user-related information, authentication, and authorization processes. It contributes to the efficient and secure operation of the network while supporting the diverse services and applications that 5G technology enables. 

Wednesday 14 March 2018

SS7(Signaling System 7 )


Signaling System 7 (it is simply called SS7) is an architecture for performing out-of-band signaling in support of the call-establishment, billing, routing and information-exchange functions of the public switched telephone network (PSTN) or public land mobile network (PLMN). It identifies functions to be performed by a signaling-system network and a protocol to enable their performance.

switched telephone network (PSTN) or public land mobile network (PLMN). It identifies functions to be performed by a signaling-system network and a protocol to enable their performance.

What is Signaling?
Signaling refers to the exchange of information between call components required to provide and maintain service.
As users of the PSTN, we exchange signaling with network elements all the time. Examples of signaling between a telephone user and the telephone network include: dialing digits, providing dial tone, accessing a voice mailbox, sending a call-waiting tone, etc.
SS7 is a means by which elements of the telephone network exchange information. Information is conveyed in the form of messages. SS7 messages can convey information such as:
  • I’ m forwarding to you a call placed from 212-555-1234 to 718-555-5678. Look for it on the trunk 067.
  • Someone just dialed 800-555-1212. Where do I route the call?
  • The called subscriber for the call on trunk 11 is busy. Release the call and play a busy tone.
  • I’m taking trunk 143 out of service for maintenance.

To initiate a call, a telephone subscriber lifts the handset off its rest-goes "off hook". This action is a signal to the exchange that the subscriber wants to make a phone call.
As soon as appropriate receiving equipment has been connected to the line, the exchange sends a dial tone back to the calling party, who then can start dialing the wanted number.
The subscriber in due course then receives advice from the exchange about the status of the call, either a ringing tone, and engaged or busy tone signal, and equipment busy tone signal (congestion), or some other specialized tone.
These are some of the signals with which the telephone subscribers themselves are concerned.
Please note that the Calling Subscriber is always referred to as the A-subscriber, and the Called Subscriber is called the B -subscriber.

If signaling is to be carried on a different path from the voice and data traffic it supports, then what should that path look like? The simplest design would be to allocate one of the paths between each interconnected pair of switches as the signaling link. Subject to capacity constraints, all signaling traffic between the two switches could traverse this link. This type of signaling is known as associated signaling.
Associated signaling works well as a switch’s only signaling requirements are between itself and other switches to which it has trunks. If call setup and management was the only application of SS7, associated signaling would meet that need simply and efficiently. In fact, much of the out-of-band signaling deployed in Europe today uses associated mode.
The North American implementers of SS7, however, wanted to design a signaling network that would enable any node to exchange signaling with any other SS7-capable node. Cleary, associated signaling becomes much more complicated when it is used to exchange signaling between nodes which do not have a direct connection. From this need, the North American SS7 architecture was born.
  
A Signaling Point (SP) is a switching or, processing node in a signaling network, with the functions of SS7 implemented.
All Signaling Points in a SS7 Signaling  Network are identified by a unique code (14 bits 0r 24 bits) known as a Signaling Point Code.
A signaling point, at which a signaling message is generated, is called the Originating Point.
A signaling point, to which a signaling message is destined, is called a Destination Point.
A signaling point, at which a message is received on one signaling link and then transferred to another link, without processing the contents of the message, is called a Signaling Transfer Point (STP).
  
The common channel signaling system uses Signaling Links (SLs) to convey the signaling messages between two signaling points.
Physically, a Signaling Link consists of a Signaling Terminal at each end of the link and some kind of transmission media (normally a time slot in a PCM -link) interconnecting the two Signaling Terminals.
A number of parallel signaling links that directly interconnect two signaling points constitute a Signaling Link Set.
  
SSPs are switches that originate, terminate, or tandem calls. An SSP sends signaling messages to other SSPs to setup, manage, and release voice circuits required to complete a call. An SSP may also send a query message to a centralized database (an SCP) to determine how to route a call (e.g., a toll-free 1-800/888 call in North America). An SCP sends a response to the originating SSP containing the routing number(s) associated with the dialed number. An alternate routing number may be used by the SSP if the primary number is busy or the call is unanswered within a specified time. Actual call features vary from network to network and from service to service.
Network traffic between signaling points may be routed via a packet switch called an STP. An STP routes each incoming message to an outgoing signaling link based on routing information contained in the SS7 message. Because it acts as a network hub, an STP provides improved utilization of the SS7 network by eliminating the need for direct links between signaling points. An STP may perform global title translation, a procedure by which the destination signaling point is determined from digits present in the signaling message (e.g., the dialed 800 number, calling card number, or mobile subscriber identification number). An STP can also act as a "firewall" to screen SS7 messages exchanged with other networks.
Because the SS7 network is critical to call processing, SCPs and STPs are usually deployed in mated pair configurations in separate physical locations to ensure network-wide service in the event of an isolated failure. Links between signaling points are also provisioned in pairs. Traffic is shared across all links in the linkset. If one of the links fails, the signaling traffic is rerouted over another link in the linkset. The SS7 protocol provides both error correction and retransmission capabilities to allow continued service in the event of signaling point or link failures.

Message Transfer Part
The Message Transfer Part (MTP) is divided into three levels. The lowest level, MTP Level 1, is equivalent to the OSI Physical Layer. MTP Level 1 defines the physical, electrical, and functional characteristics of the digital signaling link. Physical interfaces defined include E-1 (2048 kb/s; 32 64 kb/s channels), DS-1 (1544 kb/s; 24 64kb/s channels), V.35 (64 kb/s), DS-0 (64 kb/s), and DS-0A (56 kb/s).
MTP Level 2 ensures accurate end-to-end transmission of a message across a signaling link. Level 2 implements flow control, message sequence validation, and error checking. When an error occurs on a signaling link, the message (or set of messages) is retransmitted. MTP Level 2 is equivalent to the OSI Data Link Layer.
MTP Level 3 provides message routing between signaling points in the SS7 network. MTP Level 3 re-routes traffic away from failed links and signaling points and controls traffic when congestion occurs. MTP Level 3 is equivalent to the OSI Network Layer.
ISDN User Part (ISUP)
The ISDN User Part (ISUP) defines the protocol used to set-up, manage, and release trunk circuits that carry voice and data between terminating line exchanges (e.g., between a calling party and a called party). ISUP is used for both ISDN and non-ISDN calls. However, calls that originate and terminate at the same switch do not use ISUP signaling.
Telephone User Part (TUP)
In some parts of the world (e.g., China, Brazil), the Telephone User Part (TUP) is used to support basic call setup and tear-down. TUP handles analog circuits only. In many countries, ISUP has replaced TUP for call management.
Signaling Connection Control Part (SCCP)
SCCP provides connectionless and connection-oriented network services and global title translation (GTT) capabilities above MTP Level 3. A global title is an address (e.g., a dialed 800 number, calling card number, or mobile subscriber identification number) which is translated by SCCP into a destination point code and subsystem number. A subsystem number uniquely identifies an application at the destination signaling point. SCCP is used as the transport layer for TCAP-based services.
Transaction Capabilities Applications Part (TCAP)
TCAP supports the exchange of non-circuit related data between applications across the SS7 network using the SCCP connectionless service. Queries and responses sent between SSPs and SCPs are carried in TCAP messages. For example, an SSP sends a TCAP query to determine the routing number associated with a dialed 800/888 number and to check the personal identification number (PIN) of a calling card user. In mobile networks (IS-41 and GSM), TCAP carries Mobile Application Part (MAP) messages sent between mobile switches and databases to support user authentication, equipment identification, and roaming.
Operations, Maintenance and Administration Part (OMAP) and ASE
OMAP and ASE are areas for future definition. Presently, OMAP services may be used to verify network routing databases and to diagnose link problems.

What goes over signaling link?
Signaling information is passed over the signaling link in messages, which are called signal units (SUs).
Three types of SUs aredefined in the SS7 protocol.
  • Message Signal Units (MSUs)
  • Link Status Signal Units (LSSUs)
  • Fill-In Signal Units (FISUs)
SUs are transmitted continuously in both directions on any link that is in service. A signaling point that does not have MSUs or LSSUs to send will send FISUs over the link. The FISUs perform the function suggested by their name; they fill up the signaling link until there is a need to send purposeful signaling. They also facilitate link transmission monitoring and the acknowledgement of other SUs.
All transmission on the signaling link is broken up into 8-bit bytes, referred to as octets. SUs on a link are delimited by a unique 8-bit pattern know as a flag. The flag is defined as the 8-bit pattern “01111110”. Because of the possibility that data within an SU would contain this pattern, bit manipulation techniques are used to ensure that the pattern does not occur within the message as it is transmitted over the link. (The SU is reconstructed once it has been taken off the link, and any bit manipulation is reversed.) Thus, any occurrence of the flag on the link indicates the end of one SU and the beginning of another. While in theory two flags could be placed between SUs (one to mark the end of the current message and  one to mark the start of next message), in practice  a single flag is used for both purposes.
  
FISUs themselves have no information payload. Their purpose is to occupy the link at those times when there are no LSSUs or MSUs to send. Because they undergo error checking, FISUs facilitate the constant monitoring of link quality in the absence of signaling traffic. FISUs also can be used to acknowledge the receipt of messages using the backwards sequence number (BSN) and backwards indicator bit (BIB).
  
LSSUs are used to communicate information about the signaling link between the nodes on either end of the link. This information is contained in the status field of the SU. Because the two ends of a link are controlled by independent processors, there is a need to provide a means for them to communicate. LSSUs provide the means for performing this function. LSSUs are used primarily to signal the initiation of link alignment, the quality of received signaling traffic, and the status of the processors at either end of the link. Because they are sent only between the signaling points at either end of the link, LSSUs do not require any addressing information.


MSUs are the workhorses of the SS7 network. All signaling associated with call setup and tear down, database query and response , and SS7 network management takes place using MSUs. MSUs are the basic envelope within which all addressed signaling information is placed.
The functionality of the message signal unit lies in the actual content of the service information octet and the signaling information field.
  
The FIB is used in error recovery like the BIB. When a signal unit is ready for transmission, the signaling point increments the FSN (forward sequence number) by 1 (FSN = 0..127). The CRC (cyclic redundancy check) checksum value is calculated and appended to the forward message. Upon receiving the message, the remote signaling point checks the CRC and copies the value of the FSN into the BSN of the next available message scheduled for transmission back to the initiating signaling point. If the CRC is correct, the backward message is transmitted. If the CRC is incorrect, the remote signaling point indicates negative acknowledgment by toggling the BIB prior to sending the backward message. When the originating signaling point receives a negative acknowledgment, it retransmits all forward messages, beginning with the corrupted message, with the FIB toggled.
Because the 7-bit FSN can store values between zero and 127, a signaling point can send up to 128 signal units before requiring acknowledgment from the remote signaling point. The BSN indicates the last in-sequence signal unit received correctly by the remote signaling point. The BSN acknowledges all previously received signal units as well. For example, if a signaling point receives a signal unit with BSN = 5 followed by another with BSN = 10 (and the BIB is not toggled), the latter BSN implies successful receipt of signal units 6 through 9 as well.


The SIO field in an MSU contains the 4-bit SubService Field followed by the 4-bit Service Indicator. FISUs and LSSUs do not contain an SIO.
The SubService Field contains the network indicator (e.g., national or international) and the message priority (0..3 with 3 being the highest priority). Message priority is considered only under congestion conditions, not to control the order in which messages are transmitted. Low priority messages may be discarded during periods of congestion. Signaling link test messages receive a higher priority than call setup messages. Message priority, however,  is not implemented in most SS7 networks except in American .
The service indicator specifies the MTP user, thereby allowing the decoding of the information contained in the SIF.
  
In ITU-T implementations, the SLS is interpreted as the Signaling Link Code (SLC) in MTP messages.
In ITU-T Telephone User Part message, a portion of the Circuit Identification Code (CIC) is stored in the SLS field.
The Routing Label which is used by the MTP to route the messages to the correct destination contains 4 different fields:
Destination Point Code (DPC)
DPC is the part of the Label which uniquely identifies the Signaling Point to where the MSU is addressed.
Originating Point Code (OPC)
OPC is the part of the Label which uniquely identifies the Signaling Point that originates the message.
Circuit Identification Code (CIC)
CIC is the part of the Label that uniquely identifies a telephone or data circuit between the originating and the destination point.
Signaling Link Selection (SLS)
SLS is the 4 least significant bits of the CIC field. The SLS field is used to select a Signaling Link from a Signaling Link Set, normally on a load sharing basis.
Heading Codes
Each TUP message also contains an octet (8 bits) with the two Heading Codes, which uniquely identifies the type of telephone signal.
The rest of the SIF field contains a number of sub-fields (parameters) with the signaling information.

The telephone signals are transferred in the signaling network the form of signaling messages. which is the contents in the SIF field in the Message Signal Units (MSU).
The TUP signaling messages are grouped into a number of message groups, where each group is identified by a Heading Code H0. See Heading Code allocation table in this slide.
Each signaling message within a message group is identified by another Heading Code H1.
The detailed description of the TUP signals are found in the CCITT Recommendation Q.723.

FAM group
IAM    Initial Address Message
IAI      Initial Address message with additional Information
SAM   Subsequent-Address Message
SAO    Subsequent-Address message with One signal
FSM group
GSM   General forward Set-up Information Message
BSM group
GRQ   General Request Message
SBM group
ACM   Address Complete Message
UBM group
ADI     Address Incomplete message
UNN   Unallocated-National-Number signal
CSM group
ANC   Answer signa,l Charge
ANN   Answer signal, No Charge
CBK    Clear-Back signal
CLF     Clear-Forward signal
  
H0 = 0001 ---> FAM - Forward Address Message
H1 = 0001 ---> IAM - Initial Address Message
Calling Party Category (A -category)
000010 ---> operator, English
001010 ---> ordinary subscriber
Message Indicators
XXXXX1000111 ---> International number, one satellite circuit in connection, continuity check not required, outgoing half-echo suppresser included
Number of address signals ---> Number of digits in the address field
Address signals (B -number)
0011 ---> digit 3
1100 ---> digit 12

Each ISUP message contains a mandatory fixed part containing mandatory fixed-length parameters. Sometimes the mandatory fixed part is comprised only of the message type field. The mandatory fixed part may be followed by the mandatory variable part and/or the optional part. The mandatory variable part contains mandatory variable-length parameters. The optional part contains optional parameters which are identified by a one-octet parameter code followed by a length indicator ("octets to follow") field. Optional parameters may occur in any order. If optional parameters are included, the end of the optional parameters is indicated by an octet containing all zeros.

1. When a call is placed to an out-of-switch number, the originating SSP transmits an ISUP initial address message (IAM) to reserve an idle trunk circuit from the originating switch to the destination switch (1a). The IAM includes the originating point code, destination point code, circuit identification code (circuit "5" in this slide), dialed digits and, optionally, the calling party number and name. In this example, the IAM is routed via the home STP of the originating switch to the destination switch (1b). Note that the same signaling link(s) are used for the duration of the call unless a link failure condition forces a switch to use an alternate signaling link.
2. The destination switch examines the dialed number, determines that it serves the called party, and that the line is available for ringing. The destination switch rings the called party line and transmits an ISUP address complete message (ACM) to the originating switch (2a) (via its home STP) to indicate that the remote end of the trunk circuit has been reserved. The STP routes the ACM to the originating switch (2b) which rings the calling party's line and connects it to the trunk to complete the voice circuit from the calling party to the called party.
In this example, the originating and destination switches are directly connected with trunks. If the originating and destination switches are not directly connected with trunks, the originating switch transmits an IAM to reserve a trunk circuit to an intermediate switch. The intermediate switch sends an ACM to acknowledge the circuit reservation request and then transmits an IAM to reserve a trunk circuit to another switch. This processes continues until all trunks required to complete the voice circuit from the originating switch to the destination switch are reserved.
3. When the called party picks up the phone, the destination switch terminates the ringing tone and transmits an ISUP answer message (ANM) to the originating switch via its home STP (3a). The STP routes the ANM to the originating switch (3b) which verifies that the calling party's line is connected to the reserved trunk and, if so, initiates billing.
4. If the calling party hangs-up first, the originating switch sends an ISUP release message (REL) to release the trunk circuit between the switches (4a). The STP routes the REL to the destination switch (4b). If the called party hangs up first, or if the line is busy, the destination switch sends an REL to the originating switch indicating the release cause (e.g., normal release or busy).
5. Upon receiving the REL, the destination switch disconnects the trunk from the called party's line, sets the trunk state to idle, and transmits an ISUP release complete message (RLC) to the originating switch (5a) to acknowledge the release of the remote end of the trunk circuit. When the originating switch receives (or generates) the RLC (5b), it terminates the billing cycle and sets the trunk state to idle in preparation for the next call.
Note:
ISUP messages may also be transmitted during the connection phase of the call (i.e., between the ISUP Answer (ANM) and Release (REL) messages.

In the telephone services, all signaling messages and calls have some relation with the circuit. In general, the message transmission link corresponds to the call connection path.
In the GSM system,  non-circuit-related signaling message also needs to be transmitted (e.g. location update, authorization and so on), so the localization of MTP transmission emerges. What’s more, Addressing within MTP is performed according to DPC, however, the signaling point code is not the standard international code, it is only effective within a certain country. Therefore, MTP can not provide the location registration function and authorization of the international roaming subscriber.
On the other hand, the limited capacity of the signaling point code (14-bit as specified by CCITT) also limits the number of the signaling points to be marked.
And the 4-bit SI can not satisfy the increased requirements of the modern communication as well, since it can only be assigned to sixteen different user parts.
Finally, MTP can only provide the connectionless transmission. While the development of the telecommunication network needs the transmission of the large quantity of the non-realtime message, the connection has to be preset to facilitate  the connection-oriented transmission.
To solve problems described above, in 1984, the CCITT came up with a new level structure: SCCP. SCCP is based on MTP and provides the supplementary functions to MTP. When SCCP and MTP is combined together, they are called as NSP (network service part). SCCP and MTP-3 all locate in the network layer of the OSI model.
The SCCP provides additional functions to the Message Transfer Part to provide connectionless and connection-oriented network services to transfer circuit-related and non-circuit-related signaling information.
Exchange of information between two peers of the SCCP is performed by means of a protocol. The protocol is a set of rules and formats by which the control information (and user data) is exchanged between the two peers.
SCCP provides a routing function which allows signaling messages to be routed to a signaling point based on, for example, dialed digits. This capability involves a translation function which translates the global title (e.g. dialed digits) into a signaling point code and a subsystem number.
SCCP also provides a management function, which controls the availability of the "subsystems", and broadcasts this information to other nodes in the network which have a need to know the status of the "subsystem". An SCCP subsystem is an SCCP User.
Functions of the SCCP are also used for the transfer of circuit related and call related signaling information of the ISDN user part with or without setup of end-to-end logical signaling connections.

Connectionless services
Similar to the datagram transmission in the packet switching, the connectionless service does not need the preset connection (that is, signaling transmission path). SCCP may help to transmit the signaling data without establishing the signaling connection beforehand. Therefore, SCCP provides the routing function and may translate the called address into the SP code required by the MTP service.
The connectionless SCCP offers two services: The class 0 service allows the SCCP to insert SLS values randomly, or with the aim to achieve an appropriate load sharing within the underlying MTP network; The class 1 service requires the SCCP to insert the same SLS for all the SCCP-SDUs (service data unit) associated with the given parameters "sequence control" and "called address".
These two classes of the SCCP connectionless service are widely adopted in the GSM network subsystem, also be adopted in the A-interface with only class 0 service.
Connection-oriented
The connection-oriented service is similar to "virtual circuit?transmission in the data communication system. Before the message is sent, Connection-oriented services require the establishment of signaling connection (virtual connection) between the start point and the destination point via the reply mode before the transfer of signaling information by subscribers. In such a case, subscribers need not select routes by using the SCCP routing function when transferring data but transfer the data via the established signaling connection. When the transfer of data is finished, subscribers need to release the signaling connection. This type of services applies to the transfer of a big amount of data because the destination position has confirmed to be able to receive data before the data is transmitted and hence the invalid transmission of a big amount of data can be prevented. The time delay for the transmission of batches of data can be effectively curtailed.

DPC
The DPC in an address requires no translation and will merely determine if the message is destined for that SP (incoming message) or requires to be routed over the SS7 signaling network via the MTP. For outgoing messages this DPC should be inserted in the MTP routing label. The  DPC is only effective within the defined signaling network.
In the MTP routing, the SI will determine the ‘user’? e.g. TUP, SCCP, ISUP and the NI will determine which network is concerned, e.g. international or national.
SSN
SSN is the local address information employed by SCCP. It is used to distinguish each SCCP user of the same SP. For example, different SSN may be used to represent TCAP, ISUP, MAP and so on. And it will eliminate the disadvantages of the small number of the MTP message user. What’s more, it may satisfy the future development of the telecommunication service by expanding the local addressing range of the SI.
When examination of the DPC in an incoming message has determined that the message is for that SP, examination of the SSN will identify the concerned SCCP ‘user’? The presence of an SSN without a DPC will also indicate a message which is addressed to that SP.
The SSN field has an initial capacity of 255 codes with an extension code for future requirements.
GT
The Global Title (GT) is used  when the originating SP does not know the address of the destination SP, it may comprise of dialed digits or another form of addresses that will not be recognized in the SS7 network. Therefore, if the associated message requires to be routed over the SS7 network, translation is required.
Translation of the GT will result in a DPC being produced and possibly also a new SSN and GT. A field is also included in the address indicator to identify the format of the global title.

TCAP Concepts
TC – user    An application using TCAP as a protocol for communication in the network.
Dialogue    An association established between two TC -users exchanging data.
Transaction    An association between two TCAPs.
Operation    The action being requested of the remote end by a TC -user.
Component    A protocol data unit exchanged between TC -users.
TC –primitive    Primitives exchanged between TCAP and TC -user.

The 36 operations in CS-1 are:
Activate Service Filtering
Activity Test
Activity Test Response
Apply Charging
Apply Charging Report
Assist Request Instruction
Call Gap
Call Information Report
Call Information Request
Cancel Status Report Request
Collect Information
Connect
Connect to Resource
Continue
Disconnect Forward Connection
Establish Temporary Connection
Event Notification Charging
Event Report BCSM
Furnish Charging Information
Initiate DP
Initiate Call Attempt
Release Call
Request Charging Event Notification
Request Report BCSM Event
Request Status Report
Reset Timer
Select Facility
Send Charging Information
Service Filtering Response
Status Report
Assist Request Instruction from SRF
Cancel Announcement
Collected User Information
Play Announcement
Prompt and Collect User Information
Specialized Resource Report
In CS-2, the number of INAP operations has been extended to 145.



Comments are most Welcomed,

Telecom Champ Team
telecomchamp@gmail.com

Bearer Independent call control –BICC


l  What is BICC ?
an architecture that provides a means of supporting narrowband (PSTN, ISDN) services across a Packet-based backbone network without impacting the existing network interfaces and end-to-end services.
BICC is used on NC interface between two MSS. It provides interoffice call control capability independent to bearer technology in user plane & signaling transmission technology in control plane. BICC inherit all features of ISUP.

l  BICC contains the logical CIC (Call instance code) used to identify sequence of message exchanged during a call
l  a call control protocol that is unaware of the actual bearer transport being employed. Binding information identifies the bearer used for each communication instance
l  a call control protocol that is based on SS7 ISUP signalling protocol commonly used in legacy networks for PSTN/ISDN intra- and inter-networking
l  bearer (connection) control signalling protocols depend on the underlying bearer technology used (e.g., DSS2/UNI for ATM AAL type 1 and  ATM AAL type 2, IP and/or MPLS related signalling protocols)
Bearer Independent call control –BICC
BICC is modified version of ISUP that overcomes ISUP limitation to make it truly transport (bearer) independent.BICC is only standardized for ATM and IP.
BICC versions
1.       BICC CS1
2.       BICC CS2
We are using BICC capability set 2 as it allows physically separation of control servers and MGWs and MGWs selections.
BICC Principle
Concepts new to BICC as compared with ISUP
l  Call Instance Code(CIC)
l  Bearer setup direction
l  Codec negotiation
l  BICC tunneling
l  Idle bearer reuse
l  Notification
Call Instance Code(CIC):-
l  CIC is not Circuit Identification Code. BICC has the ”Call Instance Code” (CIC).The role of the “Call Instance Code” (CIC) is the identification of the signaling relation between the peer BICC entities and association of all signaling messages to that relation.
BICC Message
IAM:-
l      The BICC IAM has the same structure and format as an ISUP IAM; only BICC allows extra network specific information to be transferred. For example it could contain the identity of an already selected M-MGw or indicate the bearer setup direction or the list of supported codecs.
APM (Application Transport) :-
l      If codec negotiation is implemented, an APM message indicates to the originating exchange that a codec has been selected from the list of supported codecs.
l  Other message are ACM,ANM etc

GATEWAY CONTROL PROTOCOL (GCP/H.248/MEGACO)
The introduction of Ericsson Mobile Softswitch Solution necessitates the use of a protocol for remote control of M-MGws by control servers. The Gateway Control Protocol (GCP) was developed for this purpose. GCP operates in a master-slave configuration. Control servers, or Media Gateway Controllers (MGCs) as they are called in GCP act as masters while M-MGws act as slaves.
     MGCs issue commands to initiate connections and associations in the underlying bearer network, and may request the introduction of devices such as announcement machines, echo cancellers, DTMF devices, etc into the bearer path. M-MGws enact MGC commands and usually respond with notifications.

GCP message
ADD
    The ADD command adds a Termination to a Context. The Termination is either created or, in the case of a physical Termination, taken out of the Null Context
SUBTRACT
     The SUBTRACT command disconnect the termination from context
MODIFY
    The MODIFY command modifies Termination properties. A Termination identifier is specified if a single Termination in a Context is to be modified.
NOTIFY
     The NOTIFY command allows the M-MGw to notify the MGC of an event that has occurred.
   A Termination is a logical representation of a resource within a GCP-controlled M-MGw like malt devices. A call is through connected within a switch by associating two (or more) Terminations. A Context is an association between two or more Terminations

IP BEARER CONTROL PROTOCOL (IPBCP) :-
By introducing support for IP bearers, CN3.0 had to introduce a new signaling protocol to establish and clear down these IP bearers.It has been designed to be a tunneling protocol utilizing both the ‘vertical’ (GCP) and ‘horizontal’ (BICC) signaling protocols used to establish transport bearers. It is important to note that this protocol is only used to establish IP bearers across the Core Network (Nb interface, MGW to MGW).
      An IP bearer is a bidirectional user plane association between 2 BIWFs for carrying media stream information across IP networks. The tunneling mechanism is used to exchange media stream characteristics, port numbers and IP addresses of the source and sink of a media stream to establish. The exchange of this information is done at call establishment and after it has been established
IPBCP message
Request: the Request message is used to initiate the establishment and/or modification of an IP bearer. The iniitator of the establishment request message is known as the I-BIWF.
Accept: the Accept message is used to reply to an establish/modification request message received, only in the circumstance that it is accepted. The initiator of the accept message is known as the R-BIWF.
Confused: the confused message is sent by the R-BIWF in response to an establish/modification request message, only in the circumstance that it cannot process the received message.
Reject: the reject message is used to reply to an establish/modification request message received, only in the circumstance that it is rejection the request message received.
APM Functionality
The application using APM for bearer control is called Bearer Association Transport - Application Service Element (BAT-ASE)
APM for BICC defines among others
Action indicator (forward/backward)
BNC ID (reference used to associate the bearer with a call)
BIWF address (MGW address)
Codec(s)
Tunnelling related information (used/not used, bearer control payload)
Carried in APP parameter

Tunneling bearer information with BICC:-
IP Bearer Control Protocol (IPBCP) defines the tunneling protocol
based on SDP with BICC-specific extensions
IPBCP is carried in Bearer Control Tunneling Protocol (BCTP)
adds two bytes to indicate the used BCP
currently only IPBCP Q.1970 supported
Forward bearer setup:-
l  1) IAM conveys Application Transport parameter which has encapsulated BICC signaling information that indicates:
 - Bearer set-up is in the forward direction.
l   2) IAM sent before completion of the bearer set-up with the Nature of Connection parameter is set to indicate:
    - Continuity check performed on previous CIC
l  In addition the Application Transport parameter is conveyed with encapsulated BICC signaling information to indicate:
     - Bearer set-up is in the forward direction.
l  3) APM conveys Application Transport parameter which has encapsulated BICC signaling information to indicate:
    - Notification of bearer set-up is required.
l  4) APM conveys Application Transport parameter which has encapsulated BICC signaling information to indicate:
    - Bearer is connected
l  5) Continuity Indicators in COT indicates:
 - Successful bearer set-up on the preceding BICC leg.
Backward bearer setup:-
l  1) IAM conveys Application Transport parameter which has encapsulated BICC signaling information that indicates:
    - Bearer set-up is in the backward direction
l  2) IAM sent before completion of the bearer set-up with the Nature of Connection parameter is set to indicate:
    - Continuity check performed on previous CIC
l  In addition the Application Transport parameter is conveyed with encapsulated BICC signaling information to indicate:
    - Bearer set-up is in the backward direction.
l   3) Continuity Indicators in COT indicates:
     - Successful bearer set-up on the preceding BICC leg.
IP Bearer Control Tunneling Principle:-
l  When two MGW-s need to set up an IP bearer, they need to exchange information, i.e. the port numbers, the IP addresses, etc. The information that needs to be exchanged is defined in the IPBCP protocol.
l  When IPBCP protocol information has to be passed from one MGW to another, it is tunnelled from the MGW via the BCTP protocol over GCP using CBC extensions to the call control server, then from the call control server to the target call control server via the BICC protocol and finally from the target call control server to the target MGW via the BCTP protocol over CBC. In all this process the call control servers involved do not read or change the content of the IPBCP information.

Codec negotiation or OoBTC:-

In some cases it will be necessary to determine the codec used at each edge of the connectivity layer before the bearer is established.

IAM message contains the list of codecs supported by ingress MGW in preferred order. At the egress MGW
The highest supported codec is chosen. The chosen codec is returned in BICC APM message.
Codec negotiation is often referred as OoBTC i.e Out of Band Transcoder Control.
OoBTC is prerequisite for compressed speech in core network,TrFO and TFO.

Comments are most Welcomed,

Telecom Champ Team
telecomchamp@gmail.com

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