S UJET 1 C OMPARE SS7 AND SIP B ASED S IGNALING. M ULOT Florence S TEMMER Séverine. T HE S UBJECT. C OMPARE SS7 AND SIP B ASED S IGNALING :
COMPARE SS7 AND SIP BASEDSIGNALING
COMPARESS7AND SIP BASED SIGNALING :
Make a presentation (Power Point) which will compare SS7 and DSS1 with SIP or H.232 signaling. Use Internet, earlier presentation on www.comtel.cz, other relevant materials. Focus on using SIP or H.232 to provide telco services (voice calling with other supplementary services like CLIP, CLIR, CW, CFU, CFNR, CFBU, HOLD,...). Discuss differences and and mature of IP based signaling.
First, we used Internet to define the different words of the subject
We found different sources that we mixed together
WHAT IS SIGNALING ?
As users of the PSTN (Public Switched Telephone Network), 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, dialing *66 (to retry a busy number), …
SS7 : SIGNALING SYTEM
Signaling System #7 (SS7) is a set of telephony signaling protocols which are used to set up the vast majority of the world's PSTN (Public Switched Telephone Network) telephone calls.
It is usually abbreviated to SS7 while in North America it is often referred to as CCS7, acronym for "Common Channel Signaling System 7". In some European countries, specifically the United Kingdom, it is sometimes called C7 (CCITT number 7) and is also known as number 7 and CCIS7.
USES OF SS7
SIP : SESSION INITIATION PROTOCOL
Session Initiation Protocol is a network communications protocol commonly employed for Voice over IP (VoIP) signaling. In VoIP networking, SIP is an alternative approach to signaling using the H.323 protocol standards.
SIP is designed to support the calling features of traditional telephone systems. SIP is a peer-to-peer protocol. SIP is also a general-purpose protocol for multimedia communications not limited to voice applications.
The Real-Time Transport Protocol (RTP) used to carry the media stream does not traverse NAT routers. Most SIP clients can use STUN to traverse full cone, restricted cone, and port restricted cone NAT but not symmetrical NAT. Also some newer routers now recognize and pass SIP traffic. RTP Proxies, special purpose SIP line speed processors analogous to HTTP proxies commonly used in the early 1990s, enable CALEA and traversal of older, SIP-unaware NAT devices.
H.323 is a protocol standard for multimedia communications. H.323 was designed to support real-time transfer of audio and video data over packet networks like IP. The standard involves several different protocols covering specific aspects of Internet telephony. The International Telecommunication Union (ITU-T) maintains H.323 and these related standards.
Most voice over IP (VoIP) applications utilize H.323. H.323 supports call setup, teardown and forwarding/transfer. Architectural elements of a H.323 based system are Terminals, Multipoint Control Units (MCUs), Gateways, an optional Gatekeeper and Border Elements. Different functions of H.323 run over either TCP or UDP.
Digital Subscriber Signalling System No. 1 (DSS1), also known as Euro-ISDN or E-DSS1 (European DSS1), is a digital signaling protocol (D channel protocol) used for the ISDN. The interface is also called NET3 for BRI and NET5 for PRI lines. Can also be called CTR4.
DSS1 is a pan-European standard developed by ETSI. In 1989, 26 network operators from 20 European countries decided to develop this standard to replace earlier national protocols (such as FTZ 1 TR 6 or VN3). DSS1 has been one of the keys to the success of the ISDN within European countries (as compared to, for example, the U.S.).
Non-European countries using DSS1 include Australia, Brazil, India, Israël, New Zealand, Pakistan, Peru, Singapore, Taïwan and the United Arab Emirates.
WHAT WE HAVE LEARNT