OpenSS7, SCTP and SIGTRAN

OpenSS7 is an opensource development project to provide robust and GPL’ed SS7 stack for Linux and other UNiX operation systems.

Perhaps we should have called it LinuxSS7, or SS7-For-The-Common-Man, or SS7-For-The-Rest-Of-Us, but we’re kinda attached to the name OpenSS7 as this is an opensource project.

Project Purpose

The purpose of the OpenSS7 project is to attempt to address the following impediments to the widespread use of SS7 both inside and outside the carrier community: Expense, Complexity, Collaboration, Certification, Core Competency and Expertise.

Project Background

Lists and describes some of the significant turning points in the OpenSS7 Project. These are just the highlights.

Project Mandate

Mandate of the OpenSS7 Project: build an SS7 stack.

Project Scope

Lists and describes what components are considered within the scope of the project and which are not.

Project Objectives

Lists and describes the specific objectives of the OpensSS7 Project.

What’s New in This Release:

· The SCTP, STREAMS, and ISDN components were updated.

http://www.openss7.org

Major features since the last public release are as follows:

- Support build on openSUSE 10.2.
- Support build on Fedora 7 with 2.6.21 kernel.
- Support build on CentOS 5.0 (RHEL5).
- Support build on Ubuntu 7.04.
- Updated to gettext 0.16.1.
- Supports build on Fedora Core 6.
- Support for recent distributions and tool chains.

Asterisk utiliza este desarrollo de stack sigtran (SS7) para la comunicacion PSTN (de ser necesario).

Poryectos relacionados:

Stream Control Transmission Protocol (SCTP) is a reliable, message-oriented, multihomed transport protocol. Developed by the IETF SIGTRAN working group to transport SS7 over IP, it is now the third general-purpose transport developed by the IETF.

http://www.sctp.org/

http://www.sctp.org/implementations.html

http://www.sctp.de/

http://tdrwww.exp-math.uni-essen.de/inhalt/forschung/sctp_fb/
(SCTP for beginners)

http://sourceforge.net/projects/lksctp
(linux kernel sctp)

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. It has been designed by the IETF SIGTRAN working group, which has released the SCTP standard draft document ( RFC2960 ) in October 2000. Its design includes appropriate congestion avoidance behavior and resistance to flooding and masquerade attacks. The decisive difference to TCP is multhoming and the concept of several streams within a connection (which will be referred to as association in the rest of these documents). Where in TCP a stream is referred to as a sequence of bytes, an SCTP stream represents a sequence of messages (and these may be very short or long). SCTP can be used as the transport protocol for applications where monitoring and detection of loss of session is required. For such applications, the SCTP path/session failure detection mechanisms, especially the heartbeat, will actively monitor the connectivity of the session. An SCTP association generally looks like this, so the services of SCTP are naturally at the same layer as TCP or UDP services: Diagram showing the concept of an SCTP association _____________ _____________ | SCTP User | | SCTP User | | Application | | Application | |-------------| |-------------| | SCTP | | SCTP | | Transport | | Transport | | Service | | Service | |-------------| |-------------| | |One or more ---- One or more| | | IP Network |IP address \/ IP address| IP Network | | Service |appearances /\ appearances| Service | |_____________| ---- |_____________| SCTP Node A || SCTP Node B GSM/UMTS: RANAP, BSSAP and RNSAP over Sigtran User/Application level Protocols : : MTP lvl3 TCAP SCCP,ISUP | | | | +-----------------------+ - - +------+ - +------+ - +-------+ |User Adaptation modules| | MTP | | SCCP | | MTP 3 | +-----------------------+ | lvl2 | |------| |-------| | SCTP | | | | MTP 3| | MTP 2 | +-----------------------+ |- - - | |------| |-------| | IP Transport | | MTP | | MTP 2| | MTP 1 | +-----------------------+ | lvl1 | |------| | | | | | | MTP 1| | | Network Layer (IP) | | | | | | (a) (b) (c) (d) Figure 2.4.1: equivalence of adaptation layer to replaced layer (b) User adaptation layer = MTP lvl2 user adaptation layer (M2UA) (c) " " " = SCCP user adapatation layer (SUA) (d) " " " = MTP lvl3 User adaptation layer (M3UA) BSSAP/RANAP/RNSAP SCCP MTP3 M3UA MTP3-B MTP2 SCTP SAAL MTP1 IP ATM Stack TDM Stack IP Stack ATM

MGCP y MEGACO

Definiciones:

MGCP: Media Gateway Control Protocol (MGCP) is a protocol used within a distributed Voice over IP system.

MGCP is defined in RFC 3435, which obsoletes an earlier definition in RFC 2705. It superseded the Simple Gateway Control Protocol (SGCP).

Another protocol for the same purpose is Megaco, a co-production of IETF (RFC 3525) and ITU (Recommendation H.248-1). Both protocols follow the guidelines of the API Media Gateway Control Protocol Architecture and Requirements at RFC 2805.

Es un protocolo de séñalización en la telefonía por IP diseñado por la IEFT. MGCP fue el protocolo original, que evolucionó en MEGACO. Ambos protocolos están diseñados para la implementación en teléfonos IP que son más baratos que los teléfonos H.323 o SIP.

Está compuesto por:

• un MGC, Media Gateway Controller
• uno o más MG, Media Gateway
• uno o más SG, Signaling Gateway.

Un gateway tradicional, cumple con la función de ofrecer conectividad y traducción entre dos redes diferentes e incompatibles como lo son las de Conmutación de Paquetes y las de Conmutación de Circuitos. En esta función, el gateway realiza la conversión del flujo de datos, y además realiza también la conversión de la señalización, bidireccionalmente.

MGCP separa conceptualmente estas funciones en los tres elementos previamente señalados. Así, la conversión del contenido multimedia es realizada por el MG, el control de la señalización del lado IP es realizada por el MGC, y el control de la señalización del lado de la red de Conmutación de Circuitos es realizada por el SG.

MGCP introduce esta división en los roles con la intención de aliviar a la entidad encargada de transformar el audio para ambos lados, de las tareas de señalización, concentrando en el MGC el procesamiento de la señalización.

El control de calidad de servicio QoS se integra en el gateway GW o en el controlador de llamadas MGC. Este protocolo tiene su origen en el SGCP (de Cisco y Bellcore) e IPDC. Bellcore y Level3 plantearon el MGCP a varios organismos.

MGCP packets are unlike what you find in many other protocols. Usually wrapped in UDP port 2427, the MGCP datagrams are formatted with whitespace, much like you would expect to find in TCP protocols. An MGCP packet is either a command or a response.

Commands begin with a four-letter verb. Responses begin with a three number response code.

There are eight (8) command verbs:

AUEP, AUCX, CRCX, DLCX, MDCX, NTFY, RQNT, RSIP

AUEP - Audit Endpoint
AUCX - Audit Connection

CRCX - Create Connection
DLCX - Delete Connection
MDCX - Modify Connection

RQNT – Request for Notification

NTFY - Notify

RSIP - Restart In Progress

SIP and mobility

http://www.sipcenter.com/sip.nsf/html/SIP+and+Mobile

The services that are enabled by SIP are equally applicable in the world of mobile. A prime example is presence: a user registers their location with a SIP server and the server then knows if the user is available and where the user can be found. Location could be home, work or mobile. The mobile component, then, is crucial if services are to be portable across platforms.

As mobile networks evolve, they are becoming more concerned with data and are consequently increasingly IP-centric. With this use of IP comes SIP.

Two major technologies that impact these developments are:

• 3G and SIP
• WAP and SIP (and WAP telephony application WTA)

http://www.argreenhouse.com/sip-mobile/

http://www.cs.columbia.edu/sip/

3GPP

Release 5 and later use SIP in its Internet Multimedia Subsystem (IMS). The current status list is a good summary of the drafts, while specs/latest contains the most current drafts. In particular, TS 23.002 (architecture), TS 23.228 (concepts and procedures), 24.228 (call flows) and TS 24.229 (Call Control Protocol) contain most of the SIP-related material.

OpenSS7

OpenSS7: http://www.openss7.org/index.html

Sigtran: http://tools.ietf.org/html/rfc2719

SS7 Protocol Suite: http://www.protocols.com/pbook/ss7.htm