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Session Initiation Protocol

Session Initiation Protocol
SIP works in conjunction with several other application layer protocols that identify and carry the session media. Media identification and negotiation is achieved with the Session Description Protocol (SDP). For the transmission of media streams (voice, video) SIP typically employs the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP). For secure transmissions of SIP messages, the protocol may be encrypted with Transport Layer Security (TLS). History[edit] The protocol was designed with the vision to support new multimedia applications. SIP is distinguished by its proponents for having roots in the Internet community rather than in the telecommunications industry. Protocol operation[edit] SIP employs design elements similar to the HTTP request/response transaction model.[8] Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. Network elements[edit] User agent[edit] Registrar[edit] Related:  Asterisk7. OSI Application Layer5, 6 & 7 IPS Application Layer

[Tuto] Asterisk: Installer et configurer Asterisk sous Debian 6 et Ubuntu » Denis Rosenkranz inShare Dans ce tutoriel nous allons mettre en place un serveur de VOIP Asterisk sur un serveur linux Debian ou Ubuntu et passer un premier appel entre deux utilisateurs. Ceci n’est pas Asterisk Asterisk est un projet démarré en 1999 par Mark Spencer. La VoIP sur Asterisk passe entre autre par la prise en charge d’un protocole standard, ouvert et très largement utilisé, le SIP (Session Initiation Protocol). D’un point de vue fonctionnalité, Asterisk permet tout ce que l’on peut attendre d’un PABX moderne: La gestion des postes téléphonique sur IP locaux. Boîtes voales, transfert d’appel, mise en attente etc… Nous allons déja voir dans ce tutoriel comment installer Asterisk et le configurer pour passer un premier appel entre deux utilisateurs. Il y a deux façons d’installer Asterisk sur une distribution à base de Debian, la première via le gestionnaire de paquet de Debian, la seconde en compilant directement la dernière version d’Asterisk. Préparation à l’installation Installation Attention !

Network News Transfer Protocol As local area networks and Internet participation proliferated, it became desirable to allow newsreaders to be run on personal computers connected to local networks. Because distributed file systems were not yet widely available, a new protocol was developed based on the client-server model. It resembled the Simple Mail Transfer Protocol (SMTP), but was tailored for exchanging newsgroup articles. A newsreader, also known as a news client, is a software application that reads articles on Usenet, either directly from the news server's disks or via the NNTP. The well-known TCP port 119 is reserved for NNTP. In October 2006, the IETF released RFC 3977 which updates the NNTP protocol and codifies many of the additions made over the years since RFC 977. Network News Reader Protocol[edit] During an abortive attempt to update the NNTP standard in the early 1990s, a specialized form of NNTP intended specifically for use by clients, NNRP, was proposed. See also[edit] External links[edit]

Real Time Streaming Protocol The Real Time Streaming Protocol (RTSP) is a network control protocol designed for use in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between end points. Clients of media servers issue VCR-style commands, such as play and pause, to facilitate real-time control of playback of media files from the server. The transmission of streaming data itself is not a task of the RTSP protocol. RTSP was developed by RealNetworks, Netscape[1] and Columbia University, with the first draft submitted to IETF in 1996.[2] It was standardized by the Multiparty Multimedia Session Control Working Group (MMUSIC WG) of the Internet Engineering Task Force (IETF) and published as RFC 2326 in 1998.[3] RTSP 2.0 is currently under development as a replacement of RTSP 1.0. RTSP using RTP and RTCP allows for the implementation of rate adaptation. Protocol directives[edit] Presented here are the basic RTSP requests.

XMPP Not to be confused with XAMPP, a web server software stack. Extensible Messaging and Presence Protocol (XMPP) is a communications protocol for message-oriented middleware based on XML (Extensible Markup Language).[1] It enables the near-real-time exchange of structured yet extensible data between any two or more network entities.[2] Originally named Jabber,[3] the protocol was developed by the Jabber open-source community in 1999 for near real-time instant messaging (IM), presence information, and contact list maintenance. Designed to be extensible, the protocol has been used also for publish-subscribe systems, signalling for VoIP, video, file transfer, gaming, the Internet of Things (IoT) applications such as the smart grid, and social networking services. The Internet Engineering Task Force (IETF) formed an XMPP working group in 2002 to formalize the core protocols as an IETF instant messaging and presence technology. History[edit] Strengths[edit] Decentralization Open standards History

[Tuto] De la téléphonie IP très facilement avec Asterisk | Scrat iPhone Bonjour à tous et bonne année 2014 ! Je vous propose en ce début d’année un tuto assez simple sur Asterisk. Si vous ne le connaissez pas, c’est un logiciel libre qui permet de faire de la téléphonie par IP, c’est assez sympa à faire ! Le tutorial a été réalisé sur une Debian 7.3 64bits mais peut être appliqué sur d’autres versions (un raspberry pi par exemple). Le but à atteindre : Avoir un serveur de téléphonie fonctionnel, compilé à la main avec des pluginsPouvoir rajouter facilement des « extensions » (numéros de téléphone)Chaque utilisateur a une boite vocale personnelleLes appels groupés en conférence sont possibles Les appels pourront même être passés depuis un iPhone / Android ! Je vous avoue qu’Asterisk et moi ça n’a jamais vraiment été une grande histoire d’amour, néanmoins je pense avoir réussi à le configurer proprement et à avoir une configuration de base qui peut servir à d’autres projets plus élaborés. C’est parti ! nano /etc/network/interfaces DAHDI has been configured !

Simple Sensor Interface protocol The SSI ("Simple Sensor Interface") protocol is a simple communications protocol designed for data transfer between computers or user terminals and smart sensors. The SSI protocol is an Application layer protocol as in the OSI model. The SSI protocol has been developed jointly by Nokia, Vaisala, Suunto, Ionific, Mermit and University of Oulu. Currently SSI is being developed within the Mimosa project, part of the European Union Framework Programmes for Research and Technological Development. The SSI protocol is used in point-to-point communications over UART and networking nanoIP applications. SSI also provides polling sensors and streaming sensor data. The criteria for SSI protocol development are: general purposesimple – minimal overheadsmall footprint on the server (sensor) side Sample implementation of the SSI protocol for MSP430 microcontrollers will be published as open source during August 2006 by Nokia. SSI message structure[edit] SSI v1.2 command base[edit] The group of commands

Real-time Transport Protocol The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features. RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number. RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as RFC 1889, superseded by RFC 3550 in 2003. Overview[edit] RTP is designed for end-to-end, real-time, transfer of stream data. Protocol components[edit] [edit]

Peer-to-peer SIP Peer-to-peer SIP (P2P-SIP) is the implementation of a distributed Voice over Internet Protocol (VoIP) or instant messaging communications application using a peer-to-peer (P2P) architecture in which session or call control between communication end points is facilitated with the Session Initiation Protocol (SIP). In a pure peer-to-peer application architecture no central servers are required,[1] whereas traditional SIP telephony networks have relied on the client–server model of computing[2] using centrally deployed and managed SIP servers.[3] P2P application design can improve scalability[4] and survivability in the event of central network outages. Based on these inherent SIP features it is possible to construct a peer-to-peer network of SIP nodes (P2P over SIP). See also[edit] References[edit] External links[edit] A code library for P2P over SIP: Sip2Peer project.

Serveur OpenVPN en Monde Ponté (Bridged) Sous Debian Wheezy - Déboires d'un g33k. Nous allons voir ici la création d’un serveur OpenVpn en mode ponté, c’est-à-dire qu’à chaque client du VPN sera attribué une adresse du réseau local côté serveur. C’est le routeur en charge du DHCP côté serveur qui distribuera les adresses aux clients VPN. La carte réseau physique (eth0) et la carte réseau virtuelle (tap0) seront donc bridgées ensemble sous l’interface br0. Nous allons aussi faire en sorte de gérer la révocation des certificats. Ce tuto est en très grande partie de celui de Mattotop sur le forum debian-fr.org. Comme le paquet openssl est requis, et que tout le monde parle de Heartbleed en ce moment, on en profite pour vérifier que’on a bien la ligne deb wheezy/updates main contrib non-free dans notre sources.list :) Commencer par installer les paquets nécessaires : aptitude install bridge-utils openvpn openssl rcconf On se positionne dans le répertoire de configuration d’OpenVPN : cd /etc/openvpn On rentre dans le dossier easy-rsa : cd easy-rsa . . .

Domain Name System The Domain Name System (DNS) is a hierarchical distributed naming system for computers, services, or any resource connected to the Internet or a private network. It associates various information with domain names assigned to each of the participating entities. Most prominently, it translates domain names, which can be easily memorized by humans, to the numerical IP addresses needed for the purpose of computer services and devices worldwide. The Domain Name System distributes the responsibility of assigning domain names and mapping those names to IP addresses by designating authoritative name servers for each domain. The Domain Name System also specifies the technical functionality of the database service which is at its core. The Internet maintains two principal namespaces, the domain name hierarchy[1] and the Internet Protocol (IP) address spaces.[2] The Domain Name System maintains the domain name hierarchy and provides translation services between it and the address spaces.

Simple Network Management Protocol Simple Network Management Protocol (SNMP) is an Internet-standard protocol for collecting and organizing information about managed devices on IP networks and for modifying that information to change device behavior. Devices that typically support SNMP include cable modems, routers, switches, servers, workstations, printers, and more.[1] SNMP is widely used in network management for network monitoring. Three significant versions of SNMP have been developed and deployed. SNMP is a component of the Internet Protocol Suite as defined by the Internet Engineering Task Force (IETF). Overview and basic concepts[edit] Principle of SNMP Communication In typical uses of SNMP, one or more administrative computers called managers have the task of monitoring or managing a group of hosts or devices on a computer network. An SNMP-managed network consists of three key components: An agent is a network-management software module that resides on a managed device. Management information base[edit] GetRequest

Do not use XMPP SIP with NAT or Firewalls 1.2.1.: sip.conf port= -> The port used by asterisk for the signaling (default=5060) Bindaddr= -> The ip address on the machine asterisk has to bind to, put 0.0.0.0 to bind to all ports. Externip= -> This is an option that has to be set in the [general] context at sip.conf and has to be set to either an ip or a hostname (pointing to the external ip on your NAT device). e.g: externip=123.123.123.123 It will set the IP address in the sip address to the external ip instead of the internal IP. Localnet= -> This is an option has to be set in the [general] context at sip.conf and has to be set to the netmask for the private network asterisk is in, this is only needed when asterisk is behind a NAT and trying to communicate with devices outside of the NAT. e.g: localnet=192.168.0.0/255.255.255.0 Nat= ->This option determines the type of setting for users trying to connect to an asterisk server. a) NAT=Yes, true, y, t, 1, on b) Nat=route: c) NAT=rfc3581 d) NAT=never Qualify= a) Qualify=yes or qualify=0 rtpend=

File Transfer Protocol FTP is built on a client-server architecture and uses separate control and data connections between the client and the server.[1] FTP users may authenticate themselves using a clear-text sign-in protocol, normally in the form of a username and password, but can connect anonymously if the server is configured to allow it. For secure transmission that hides (encrypts) the username and password, and encrypts the content, FTP is often secured with SSL/TLS ("FTPS"). SSH File Transfer Protocol ("SFTP") is sometimes also used instead, but is technologically different. History[edit] The original specification for the File Transfer Protocol was written by Abhay Bhushan and published as RFC 114 on 16 April 1971. Until 1980, FTP ran on NCP, the predecessor of TCP/IP.[2] The protocol was later replaced by a TCP/IP version, RFC 765 (June 1980) and RFC 959 (October 1985), the current specification. Protocol overview[edit] Communication and data transfer[edit] ASCII mode: used for text. Login[edit] or:

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