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:
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]
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
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] Free SIP and XMPP accounts for voice, video, IM and presence | getonsip Seamlessly conduct audio and video calls in your Chrome or Firefox browser. There is absolutely nothing to download, and communications are peer-to-peer (secure). Invite people to communicate with your instant call link. <p>Use SIP applications on your smartphone and take calls from anywhere. Use SIP applications on your smartphone and take calls from anywhere. GetOnSIP works with downloadable soft phone applications on your PC or laptop. Register your SIP address on your home or office phone for free voice and video via SIP.
Post Office Protocol Overview[edit] POP supports simple download-and-delete requirements for access to remote mailboxes (termed maildrop in the POP RFC's).[3] Although most POP clients have an option to leave mail on server after download, e-mail clients using POP generally connect, retrieve all messages, store them on the user's PC as new messages, delete them from the server, and then disconnect. Other protocols, notably IMAP, (Internet Message Access Protocol) provide more complete and complex remote access to typical mailbox operations. In the late 90's and early 2000's, fewer Internet Service Providers (ISPs) supported IMAP due to the storage space that was required on the ISP's hardware. Contemporary e-mail clients supported POP, then over time popular mail client software added IMAP support. A POP3 server listens on well-known port 110. History[edit] POP1 was specified in RFC 918 (1984), POP2 by RFC 937 (1985) POP3 originated with RFC 1081 (1988). Extensions[edit] STARTTLS[edit] SDPS[edit] [edit]
BroadcastReceiver Base class for code that will receive intents sent by sendBroadcast(). If you don't need to send broadcasts across applications, consider using this class with LocalBroadcastManager instead of the more general facilities described below. This will give you a much more efficient implementation (no cross-process communication needed) and allow you to avoid thinking about any security issues related to other applications being able to receive or send your broadcasts. There are two major classes of broadcasts that can be received: Even in the case of normal broadcasts, the system may in some situations revert to delivering the broadcast one receiver at a time. In particular, for receivers that may require the creation of a process, only one will be run at a time to avoid overloading the system with new processes. The BroadcastReceiver class (when launched as a component through a manifest's <receiver> tag) is an important part of an application's overall lifecycle. Topics covered here: Security
Telnet Telnet is a network protocol used on the Internet or local area networks to provide a bidirectional interactive text-oriented communication facility using a virtual terminal connection. User data is interspersed in-band with Telnet control information in an 8-bit byte oriented data connection over the Transmission Control Protocol (TCP). Telnet was developed in 1968 beginning with RFC 15, extended in RFC 854, and standardized as Internet Engineering Task Force (IETF) Internet Standard STD 8, one of the first Internet standards. Historically, Telnet provided access to a command-line interface (usually, of an operating system) on a remote host. History and standards[edit] Before March 5, 1973, Telnet was an ad hoc protocol with no official definition.[1] Essentially, it used an 8-bit channel to exchange 7-bit ASCII data. Security[edit] It is of note that there are a large number of industrial and scientific devices which have only Telnet available as a communication option. Telnet 5250[edit]