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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:  5, 6 & 7 IPS Application Layer

TUTORIAL: How to create two 100% free SIP accounts to try the SIP Connector... Anyone with a broadband internet connection can try the SIP Connector. If you don't know VOIP, or if you don't have access to a SIP account, you can see how it works by following this tutorial. You are going to use the free SIP service provided by pbxes.org, in order to create 2 SIP extensions. First extension is AccountName-600 for HomeSeer use. 1) Create two 100% free SIP Accounts Go to Choose Create Account, and fill the form. Once this is done, in the left menu choose: Extensions Add extension SIP Extension Number : 600 Display Name : HomeSeer Password: xxx SUBMIT Extension Number : 601 Display Name : MyName Password: xxx SUBMIT Then click on the red bar. In the left menu choose: Incoming Calls Choose Extension HomeSeer <600> for both regular and after hours SUBMIT Then click on the red bar. 2- Download, Install X-LITE and configure for extension 601 Download from this link 3- Configure SIP Connector for extension 600

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.

libsofia-sip-ua: Sofia SIP User Agent Library - sofia-sip-ua Introduction This document contains automatically generated reference documentation for Sofia-SIP components. Some introductory material and pointers to the example code are also included. For a list of module specific pages, see source tree structure or direct links to submodules: nua su features soa sdp nta tport sresolv sip msg url stun iptsec nea nth http bnf ipt Contact Information You can download latest Sofia SIP from the project home page at Sourceforge.net. Please contact us if you have questions regarding this software: Or post to the Sofia-SIP mailing list:

XMPP Communications protocol for message-oriented middleware Extensible Messaging and Presence Protocol (XMPP, originally named Jabber[1]) is an open communication protocol designed for instant messaging (IM), presence information, and contact list maintenance.[2] Based on XML (Extensible Markup Language), it enables the near-real-time exchange of structured data between two or more network entities.[3] Designed to be extensible, the protocol offers a multitude of applications beyond traditional IM in the broader realm of message-oriented middleware, including signalling for VoIP, video, file transfer, gaming and other uses. Originally developed by the open-source community, the protocols were formalized as an approved instant messaging standard in 2004 and have been continuously developed with new extensions and features. Protocol characteristics[edit] Decentralization[edit] Addressing[edit] Since a user may wish to log in from multiple locations, they may specify a resource. Features[edit]

Buttons A button consists of text or an icon (or both text and an icon) that communicates what action occurs when the user touches it. Depending on whether you want a button with text, an icon, or both, you can create the button in your layout in three ways: With text, using the Button class: <Button android:layout_width="wrap_content" android:layout_height="wrap_content" android:text="@string/button_text" ... /> With an icon, using the ImageButton class: <ImageButton android:layout_width="wrap_content" android:layout_height="wrap_content" android:src="@drawable/button_icon" ... /> With text and an icon, using the Button class with the android:drawableLeft attribute: <Button android:layout_width="wrap_content" android:layout_height="wrap_content" android:text="@string/button_text" android:drawableLeft="@drawable/button_icon" . Responding to Click Events When the user clicks a button, the Button object receives an on-click event. For example, here's a layout with a button using android:onClick:

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]

Session Description Protocol The Session Description Protocol (SDP) is a format for describing streaming media initialization parameters. The IETF published the original specification as an IETF Proposed Standard in April 1998,[1] and subsequently published a revised specification as an IETF Proposed Standard as RFC 4566 in July 2006.[2] SDP is intended for describing multimedia communication sessions for the purposes of session announcement, session invitation, and parameter negotiation. SDP does not deliver media itself but is used for negotiation between end points of media type, format, and all associated properties. The set of properties and parameters are often called a session profile. SDP started off as a component of the Session Announcement Protocol (SAP), but found other uses in conjunction with Real-time Transport Protocol (RTP), Real-time Streaming Protocol (RTSP), Session Initiation Protocol (SIP) and even as a standalone format for describing multicast sessions. Session description[edit] Notes[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.

Session Initiation Protocol Android provides an API that supports the Session Initiation Protocol (SIP). This lets you add SIP-based internet telephony features to your applications. Android includes a full SIP protocol stack and integrated call management services that let applications easily set up outgoing and incoming voice calls, without having to manage sessions, transport-level communication, or audio record or playback directly. Here are examples of the types of applications that might use the SIP API: Video conferencing. Requirements and Limitations Here are the requirements for developing a SIP application: You must have a mobile device that is running Android 2.3 or higher. SIP API Classes and Interfaces Here is a summary of the classes and one interface (SipRegistrationListener) that are included in the Android SIP API: Creating the Manifest To use SIP, add the following permissions to your application's manifest: android.permission.USE_SIP android.permission.INTERNET <uses-sdk android:minSdkVersion="9" />.

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 SipManager Class Overview The APIs in this class allows you to: Summary Constants public static final String EXTRA_CALL_ID Key to retrieve the call ID from an incoming call intent. Constant Value: "android:sipCallID" public static final String EXTRA_OFFER_SD Key to retrieve the offered session description from an incoming call intent. Constant Value: "android:sipOfferSD" public static final int INCOMING_CALL_RESULT_CODE The result code to be sent back with the incoming call PendingIntent. Constant Value: 101 (0x00000065) Public Methods public void close (String localProfileUri) Closes the specified profile to not make/receive calls. Parameters public SipSession createSipSession (SipProfile localProfile, SipSession.Listener listener) Creates a SipSession with the specified profile. public static String getCallId (Intent incomingCallIntent) Gets the call ID from the specified incoming call broadcast intent. Returns the call ID or null if the intent does not contain it the session object that handles the incoming call

Open Network Computing Remote Procedure Call ONC RPC was described in RFC 1831, published in 1995. RFC 5531, published in 2009, is the current version. Authentication mechanisms used by ONC RPC are described in RFC 2695, RFC 2203, and RFC 2623. In 2009, Sun relicensed the ONC RPC code under the standard 3-clause BSD license[1] and then reconfirmed by Oracle Corporation in 2010 following confusion about the scope of the relicensing.[2] ONC is considered "lean and mean", but has limited appeal as a generalized RPC system for WANs or heterogeneous environments[citation needed]. See also[edit] References[edit] Birrell, A. Notes[edit] External links[edit] How PGP works How PGP works The following text is taken from chapter 1 of the document Introduction to Cryptography in the PGP 6.5.1 documentation. Copyright © 1990-1999 Network Associates, Inc. and its Affiliated Companies. All Rights Reserved. Converted from PDF to HTML at and then manually edited by hand. The Basics of Cryptography When Julius Caesar sent messages to his generals, he didn't trust his messengers. And so we begin. Encryption and decryption Data that can be read and understood without any special measures is called plaintext or cleartext. Figure 1-1. What is cryptography? Cryptography is the science of using mathematics to encrypt and decrypt data. While cryptography is the science of securing data, cryptanalysis is the science of analyzing and breaking secure communication. Cryptology embraces both cryptography and cryptanalysis. Strong cryptography --Bruce Schneier, Applied Cryptography: Protocols, Algorithms, and Source Code in C. How does cryptography work? Keys

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