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Wireless Telecommunication Protocols for GSM / UMTS (2G, 3G & onwards)

Updated on March 9, 2011

Signalling Principle

The basic element for SS7 signalling information transfer is signalling link. Signalling link is a data-link-layer connection between two signalling nodes. Both nodes identify a signalling link with a unique number, signalling link code (SLC). The set of signalling links between two signalling nodes is called signalling link set (SLS). If load sharing is taken into use, the MTP level is able to distribute

messages of one signalling session over several links. The signalling node taking care of rerouting of the messages in this kind of case is called STP. In the originating signalling node, the routing entity on the MTP level is called signalling route set. Signalling route set is the collection of the signalling routes,

through which a certain SPC can be achieved. Signalling route is in practice the same as the signalling link set, but the difference here is that signalling link set is not “aware” of the STP facility, but the signalling route is.

Various mnemonics have been created over the years to help remember the order. Probably the most taught and textbook-found are:

  • Please Do Not Throw Sausage Pizza Away!
  • Please Do Not Take Sales-People's Advice

Others include:

  • Australia Post Sucks They Never Deliver Parcels
  • All Pre-School Toys Need Durable Parts
  • A "Perfect" System That Never Did look Perfect
  • All People Seem To Need Data Processing

MTP routing facility checks the Originating Signalling Point Code (OPC) and Destination Signalling Point Code (DPC) and if the DPC is the same than defined for the current MTP node, the message is interpreted to be terminated in this signalling point. If different, the message is re-routed towards the correct DPC. The SIF address part contains identification for used signalling channel, Signalling Link Selection (SLS) and circuit concerned, Circuit Identification Code (CIC).

Signalling System number 7 (SS7) describes a signalling structure and related protocols for digital exchanges. It includes several application services for different purposes over its network layer protocol SCCP:

  • ISDN User Part (ISUP) for PSTN signalling
  • RAN Application Part (RANAP) for UMTS signalling
  • BSS Application Part (BSSAP) for GSM signalling
  • Mobile Application Part (MAP) for mobility management in the MSS core network
  • CAMEL Application Part (CAP) for providing the intelligent network (IN) services
  • Intelligent Network Application Part (INAP) for IN services

2G & 3G Protocols

SS7 & Sigtran Interworking
SS7 & Sigtran Interworking


1 Base Station System Application Part (BSSAP)

Signalling over the A-interface (between the MSC and the BSS) is done according to ETSI/GSM BSSAP based on CCITT Signalling System Number 7.

2 Channel Associated Signalling (CAS)

The signalling systems used between an MSC/VLR and the PSTN are individually designed or selected for each MSC/VLR. This includes also standardized CAS systems, such as:

  • Register Signalling National R2 MFC
  • Line Signalling System based on CCITT R2 PCM

3 DSS1 Protocol

The Digital Subscriber Signalling System No. 1 (DSS 1) provides the necessary signaling procedures required for establishing, maintaining and clearing of network connections and for control of supplementary services at the ISDN user-network interface, for both basic and primary rate access.

4 Intelligent Network Protocol

The INAP protocol enables the SCF to remotely control circuit switched call processes in real time through the SSF. The INAP protocols use TCAP/SCCP signalling.

The MSC/VLR via SSF supports the following protocols:

  • CS-1: ETSI Core INAP
  • CS-1+: Ericsson enhanced Core INAP
  • CAP: CAMEL Application Part phase 1 and 2

and a subset of CAMEL phase 3.

CAMEL Customised Applications for Mobile Network Enhanced Logic

3.2.5 Mobile Application Part (MAP)

MAP is an SS7 protocol that is used to transfer subscriber data between GSM network elements, or between GSM and other system network elements. MAP protocol is not directly involved in traffic delivery. Instead, it performs a numerous supportive operations, related to handling of different identities (both static and variable) and their locations within the core network.

MAP receives internally-coded data from a MAP user, for example, from an HLR application, a VLR application, or an SMS application, and sends the data in a standardised format, that is, in the ASN.1 format, to a peer network element. MAP is needed because signalling is more complicated in the GSM network than in the PSTN. Non-call-related signalling and the mobility of subscribers make additional demands. An example of a MAP procedure is the location updating in which MAP is used for signalling between the VLR and the HLR.

The following network elements have the MAP interface:

  • Home Location Register (HLR)
  • Visitor Location Register (VLR)
  • Mobile Services Switching Centre (MSC)
  • Equipment Identity Register (EIR)
  • Serving GPRS Support Node (SGSN)
  • GSM Service Control Function (gsmSCF)
  • Gateway MobileLocationCenter (GMLC)

Nokia has implemented MAP as specified by ETSI and its successor 3GPP. Compatibility between different network elements provided by different vendors has been proven in a number of networks and cases, where Nokia has been the other equipment supplier.

· MAP-B between the MSC & the VLR

· MAP-C between the MSC & the HLR

· MAP-D between the VLR & the HLR

· MAP-E between two MSCs

· MAP-F between the MSC & the EIR

· MAP-G between two VLRs

· MAP-H between the MSC & the SMSC

· MAP-I between the subscriber & the HLR

MAP call handling procedures

The MAP call handling procedures are used to retrieve routing information to handle a mobile-terminated call, to transfer the control of a call back to the GMSC if the call is to be forwarded, to retrieve and to transfer information between the anchor MSC and the relay MSC for inter-MSC group calls/broadcast calls, to handle reporting of MS status for call completion services, and to handle notification of remote user free for Completion of Calls to Busy Subscribers (CCBS).

The following call handling procedures are used:

  • retrieval of routing information
  • transfer of call handling
  • inter-MSC group call
  • setting of reporting state
  • status reporting
  • remote user free

Support protocol description of MAP

MAP uses the services of the other signalling protocol layers of the system.

These protocol layers are:

  • MTP (Message Transfer Part)
  • SCCP (Signalling Connection Control Part)
  • TCAP (Transaction Capabilities Application Part)

MTP is the foundation on which SS7 is built.

SCCP enables the placement of signalling messages in the correct order at the distant end (connection-oriented network service) and signalling across multiple networks in the absence of a call (connectionless network service).

MAP uses only the following SCCP's connectionless services:

  • Class 0: Basic connectionless class
  • Class 1: Sequenced connectionless class

SCCP (Signalling Connection Control Part) to offer connection-oriented and connectionless services within a SS7 environment, additional protocol called Signalling Connection Control Part (SCCP) is required and this protocol sits on top of the basic SS7 protocol stack. SCCP uses Global Title addressing, which makes it possible, for example, for an SGSN within a visited network to reach the HLR in the subscriber's home network by using the GT of the HLR for addressing. The address of SCCP includes GT, SPC and SSN codes and give new GT which will be convenient for MTP.

TCAP (Transaction Capabilities Application Part) handles the MAP transaction messages between different network elements. The TCAP supports MAP and is used as a dialog handler between the databases. (HLR, VLR, etc.).

MAP uses the SCCP's services through TCAP.

MTP can be replaced by the SIGTRAN protocol stack (M3UA, SCTP, and IP) to transfer SS7 signalling over IP networks.

When one network element that has MAP, operates with another network element, both elements have to have the above-mentioned protocol stack, that is, MAP, TCAP, SCCP, and MTP.

A network element can also serve as an STP between other network elements when it only transmits signalling from one element to another. In this case, the STP can only have MTP or MTP and SCCP even if the other network elements also have MAP.

The ETSI/GSM MAP provides the necessary signalling procedures required for information exchange between the MSC and the Location Registers, and between the MSCs. MAP uses CCITT Signalling System Number 7 for transfer of the information, i.e. Transaction Capability Application Part (TCAP), Signalling Connection Control Part (SCCP) and the Message Transfer Part (MTP) of CCITT Number 7.

BSSGP (Base Station Subsystem GPRS Protocol) The primary function of BSSGP is to provide the radio-related, QoS, and routing information that is required to transmit user data between a BSS and an SGSN.

The primary functions of the base station sub-system GPRS protocol (BSSGP) include, in the downlink, the provision by an SGSN to a BSS of radio-related information used by the RLC/MAC function; in the uplink, the provision by a BSS to an SGSN of radio-related information derived from the RLC/MAC function; and the provision of functionality to enable two physically distinct nodes, an SGSN and a BSS, to operate node management control functions.

BSSAP(from 2G side) subsystem Base station subsystem application part is used to control GSM-specific serves on the An interface.

BSSAP+ Base Station Subsystem Application Part

CHAP Challenge-Handshake Authentication Protocol

GTP (GPRS tunneling protocol) The GPRS Tunnelling Protocol (GTP) is the protocol used between GPRS Support Nodes (GSNs) in the UMTS/GPRS backbone network. GTP allows multi-protocol packets to be tunnelled through the UMTS/GPRS Backbone between GSNs. The mapping of GTP tunnels to PDP contexts is done by using the Tunnel Endpoint Identifier (TEID) in the GTP header.

GTP-C (GTP for Control plane) is used on the Gn interface. GTP specifies a tunnel control and management protocol (GTP-C) which allows the SGSN to provide packet data network access for an

MS. Control Plane signalling is used to create, modify and delete tunnels. The control plane relates to GPRS Mobility Management functions like for example GPRS Attach, GPRS Routing Area Update and Activation of PDP Contexts.

GTP-U (GTP for User plane) The GTP used for the user plane is called GTP-U. This is to separate it from the GTP control plane protocol – GTP-C. GTP uses a tunnelling mechanism (GTP-U) to provide a service for carrying user data packets. GTP-U Tunnels are used to carry encapsulated T-PDUs and signalling messages between a given pair of GTP-U Tunnel Endpoints. The Tunnel Endpoint ID (TEID) which is present in the GTP header shall indicate which tunnel a particular T-PDU belongs to.

GTP’ (Enhanced GPRS Tunnelling Protocol) GTP’ uses the same header structure as GTP. In GTP' messaging only the signalling plane of GTP is partly reused. GTP' defines a set of messages between two associated nodes. Echo request/response and Version not supported are messages reused from the GTP protocol. Six other messages are defined for GTP’:

  • Node Alive Request, used to inform that a node in the network has started its service (e.g. after a service break due to software or hardware maintenance or data service interruption after an error condition).
  • Node Alive Response
  • Redirection Request, to advise that received CDR traffic is to be redirected to another CG
  • Redirection Response
  • Data Record Transfer Request, used to transmit the CDR(s) to the CG
  • Data Record Transfer Response

GRE Generic Routing Encapsulation

GMM GPRS Mobility Management Protocol

HSRP Hot Standby Routing Protocol

HTTP Hypertext Transfer Protocol

IRAP International Roaming Access Protocol

IMAP Internet Message Access Protocol

IP Internet Protocol

FTP File Transfer Protocol

L2TP Layer 2 Tunnelling Protocol

L4 Layer 4, transport layer (according to the OSI network reference model)

L7 Layer 7, application layer (according to the OSI network model)

LDAP Lightweight Directory Access Protocol

LIPv1 Lawful Interception Protocol, Nokia Proprietary

NTP Network Time Protocol

OSI Open Systems Interconnection reference model

OOB Out-Of-Box Interface; is used to pass traffic transparently to an external OOB Network element (ONE) for further analysis and manipulation. The OOB interface supports both control and user planes traffic.

OSPF Open Shortest Path First

OSPFv3 OSPF for Ipv6

PAP Password Authentication Protocol

PDP Packet Data Protocol

POP Post Office Protocol

RTP (Real Time Transport Protocol): an Internet protocol for transmitting real-time data such as audio and video. It defines a standard packet format for delivering audio and video over the internet. It is used mainly for streaming multimedia applications: live radio and television broadcast, webcast concerts, and video conferencing. It is defined in RFC 1889. It was developed by the Audio Video Transport Working group and was first published in 1996.

Typically, RTP runs on top of the UDP protocol, although the specification is general enough to support other transport protocols.

RTP and RTCP are closely linked – RTP delivers the actual data and RTCP is used for feedback on quality of service.

RTCP (Real Time Transport Control Protocol): RTCP works hand in hand with RTP. RTP does the delivery of the actual data, whereas RTCP is used to send control packets to participants in a call. The primary function is to provide feedback on the quality of service being provided by RTP.

RTSP (Real-Time Streaming Protocol) is used for controlling streaming video and audio over the Internet.

RANAP (RAN Application Part (RANAP) Protocol) from 3G side (it is used for the MS/UE attach and routing analysis.) The communication between the UE and the SGSN is tunnelled within the RANAP connection between the RNC and the SGSN. RANAP messages are used for all signalling transfer between the SGSN and the RNC.

TFO Tandem Free Operation is a technique used in GSM networks to reduce the number of speech transcodings made in the end-to-end path of the call. It investigates whether both ends of the call use the same speech codec. If this is the case, the unnecessary transcoding phases are by-passed.

TrFO (Transcoder Free Operation) is a technique used in UMTS networks to reduce the number of speech transcodings made in the end-to-end path of the call. It uses a specific codec negotiation to investigate whether both ends of the call use the same speech codec, thus permitting unnecessary transcoding phases to be by-passed and enabling the same codec to be used during the whole transmission.

RIP Routing Information Protocol

RIPng RIP for Ipv6, next generation

SOAP Simple Object Access Protocol (XML over HTTP)

SNMP Simple Network Management Protocol

SMTP Simple Mail Transfer Protocol

SCTP (Signalling Common Transport Protocol) protocol specifically designed to transport signalling protocols, which require a telecom carrier-grade quality of service, over IP networks.

(Stream Control Transmission Protocol) SCTP is a Sigtran-defined protocol that provides connection-oriented signalling streams for carrying SS7 signalling information.

SCTP (Stream Control Transmission Protocol) layers. SCTP is a Sigtran-defined protocol that provides connection-oriented signalling streams for carrying SS7 signalling information.

One of the main features of SCTP is its support for multihoming. With SCTP, several IP addresses can be defined to one (asymmetric) or both (symmetric) endpoints of the path. To reap all benefits of multihoming, the host needs multiple network interfaces, each of which must be configured to work in a different subnetwork. In failure cases the path can be automatically switched from the primary path to the secondary one.

SCTP is a reliable transport protocol operating on top of a potentially unreliable connectionless packet service such as IP. It offers acknowledged error-free non-duplicated transfer of datagrams (messages). Detection of data corruption, loss of data and duplication of data is achieved by using checksums and sequence numbers. A selective retransmission mechanism is applied to correct loss or corruption of data. Originally, SCTP was designed to provide a general-purpose transport protocol for message-oriented applications, as is needed for the transportation of signalling data. Its design includes appropriate congestion avoidance behaviour and resistance to flooding and masquerade attacks. SCTP can be used as the transport protocol for applications where monitoring and detection of loss of session is required. The SCTP ensures that none of the messages can get lost if only one path is broken at the same time.

SIP(Session Initiation Protocol) is an IETF-defined text-based protocol for initiating the user sessions that can include, for example, voice and video. Basic SIP is used to establish, modify and terminate calls or multimedia sessions between SIP client and server.

SIP-T (SIP for Telephones) provides an alternative to BICC signalling in the Nokia MSS system.

IETF SIP-T enables ISUP tunnelling, where the ISUP message is encapsulated in the SIP body. SIP-T is used in particular in the Nc interface between MSSs and enables the existing IN and ISDN services such as call-forwarding, call-transfer, call-hold, call-waiting and capability negotiation.

ITU SIP-I (SIP with encapsulated ISUP) has three different profiles:

  • SIP-I profile A conveys 3GPP specific parameters used in particular for IMS
  • SIP-I profile B is IETF variant of SIP for ISUP/BICC interworking
  • SIP-I profile C enables ISUP tunnelling, where ISUP is MIME encoded

SSL Secure Sockets Layer

SSH Secured Shell is a protocol for remotely logging into a machine via a shell.

SCP Secure CoPy, is a protocol that allows you to transmit files from one machine to another with the encryption benefits of SSH.

SRI (Send Routing info) As an option, the gsmSCF may determine the MSRN of the served subscriber by sending the SendRoutingInfo operation (including the gsmSCF Initiated Call parameter) to HLR. In the HLR this is handled in a similar manner as an SRI from a GMSC. The HLR responds with an SRI-ACK containing the roaming number. The gsmSCF can indicate to the HLR that incoming call barring, call diversion, and the VT-CSI must be suppressed for this call.

WAP Wireless Application Protocol

WDP Wireless Datagram Protocol

WSP Wireless Session Protocol

WTP Wireless Transaction Protocol

UDP (User Datagram Protocol): a connectionless protocol that, like TCP, runs on top of IP networks. Unlike TCP/IP, UDP/IP provides very few error recovery services, offering instead a direct way to send and receive datagrams over an IP network. It's used primarily for broadcasting messages over a network.

VRRP Virtual Router Redundancy Protocol

Diameter this protocol is intended to provide a framework for any services which require AAA (Access, Authorization, and Accounting)/Policy support across many networks. Access and Authentication are achieved through a key distribution system coordinated through a key broker. The main functions of DIAMETER are to support MIP (Mobile IP), Accounting, Network Access and Strong Security. The Diameter interface is based on the Diameter Credit Control Application specified by the IET.

This protocol is used in the interface between the GGSN and the Nokia Online Service Controller (OSC) to pass on charging data.


IETF working group that prepares Internet drafts which define how narrowband circuit-switched network signalling protocols can be transported over an IP network in order to support time division multiplex to IP interworking of voice and data services.

The MSC Server uses 'Sigtran' to carry SS7 signalling, i.e. RANAP, BSSAP, MAP and ISUP over IP.

In fact, Sigtran (signalling transport) is the name of the IETF's working group defining frameworks and protocols for signalling transport over IP.

IETF SIGTRAN protocols change the three lowest layers (Message Transfer Part (MTP) layers 1-3) of narrowband SS7 signalling protocol stacks, while the upper layers stay the same as with PCM based signalling. The main enabling protocol for IP-based signalling transfer is the Stream Control Transmission Protocol (SCTP). This protocol was specifically designed to support capabilities similar to those found in Message Transfer Part (MTP), but on an unreliable IP transport. SS7 signalling transport over IP contains several user adaptation layers, however, here only the MTP3 User Adaptation (M3UA) layer is used. M3UA provides an interface that is compatible with MTP3. In other words, it supports

MTP3 applications on top of SCTP and IP.

M3UA protocol which is a part of the IETF signalling transport (SIGTRAN) stack for the transport of any signalling system No.7 (SS7) MTP3-user signalling over IP using the services of the stream control transmission protocol (SCTP)

The M3UA adaptation layer provides SS7 routing information (similar to MTP3) and encapsulation of higher layer SS7 application parts into the SCTP frames.

M3UA supports the transport of any SS7 MTP3-User signalling (such as ISUP and SCCP messages) over IP, using the services of the Stream Control Transmission Protocol (SCTP). The protocol is used for communication between a Signalling Gateway (SG) and the SGSN. It is assumed that the SG receives SS7 signalling over a standard SS7 interface using the SS7 Message Transfer Part (MTP) to provide transport.

BICC The Bearer Independent Call Control protocol is used between two MSC Servers – in principle, to carry modified ISUP call control messages and optional “bearer control information”. It can be used between two MSC servers during the call setup for the bearer-independent circuit core. In the Nokia MSS System the BICC messages are typically carried over M3UA and SCTP, but theoretically BICC can also be carried over UDP, TCP, ATM or TDM connections. In all cases, user plane traffic shall have IP or ATM backbone for BICC. As an alternative to BICC, SIP can be used as a call setup protocol for IP bearers and ISUP can be used for TDM bearers.

To be accurate, BICC does not carry ISUP but 'modified application protocol' that supports modified ISUP call control messages such as Initial Address Message (IAM) or Answer Message (ANM). It also carries new messages such as Application Transport Mechanism (APM) allowing MSC Servers to interchange required bearer information regarding the call. In the Nokia MSC Server solution the BICC messages are typically carried over M3UA and SCTP, but theoretically BICC can also be carried over UDP, TCP, ATM or TDM connections.


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    • profile image

      manjit saini 

      6 years ago

      thanks so much

    • naumaan profile imageAUTHOR


      7 years ago from Dubai

      Thanks for appreciation. I included it now as well.

    • profile image

      geographic telephone numbers 

      7 years ago

      Great list of acronyms and meanings.

      However RTP is missing from the list and is an important acronym in VoIP and goes hand in hand with SIP.

      RTP - Real Time Protocol. This is used by VoIP to transmit media, usually voice, video and DTMF. It is usually carried over the UDP transport and is separate from the signalling protocol.


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