Free rtp software for linux

The SR-RTP library enables real-time streaming applications (such as streaming MPEG-4 video) to cope with packet loss, variable bandwidth, and variable delay on the Internet.

It provides a means for selective retransmission of lost packets in a fashion that is backwards-compatible with RTP. Additionally, it provides integration with the Congestion Manager to provide a system capable of performing TCP- friendly streaming of real-time data.

Installation:

The `configure shell script attempts to guess correct values for various system-dependent variables used during compilation.

It uses those values to create a `Makefile in each directory of the package. It may also create one or more `.h files containing system-dependent definitions.

Finally, it creates a shell script `config.status that you can run in the future to recreate the current configuration, a file `config.cache that saves the results of its tests to speed up reconfiguring, and a file `config.log containing compiler output (useful mainly for debugging `configure).

If you need to do unusual things to compile the package, please try to figure out how `configure could check whether to do them, and mail diffs or instructions to the address given in the `README so they can be considered for the next release.

If at some point `config.cache contains results you dont want to keep, you may remove or edit it.

The file `configure.in is used to create `configure by a program called `autoconf. You only need `configure.in if you want to change it or regenerate `configure using a newer version of `autoconf.

The simplest way to compile this package is:

1. `cd to the directory containing the packages source code and type `./configure to configure the package for your system. If youre using `csh on an old version of System V, you might need to type `sh ./configure instead to prevent `csh from trying to execute
`configure itself.

Running `configure takes awhile. While running, it prints some messages telling which features it is checking for.

2. Type `make to compile the package.

3. Optionally, type `make check to run any self-tests that come with the package.

4. Type `make install to install the programs and any data files and documentation.

5. You can remove the program binaries and object files from the source code directory by typing `make clean. To also remove the files that `configure created (so you can compile the package for a different kind of computer), type `make distclean.

There is also a `make maintainer-clean target, but that is intended mainly for the packages developers. If you use it, you may have to get all sorts of other programs in order to regenerate files that came with the distribution.

JRTPLIB is an object-oriented library written in C++ which offers support for the Real-time Transport Protocol (RTP) as defined in RFC 3550.
JRTPLIB project makes it very easy to send and receive RTP packets and the RTCP (RTP Control Protocol) functions can be handled entirely internally.
Enhancements:
- A memory management system was added.
- A bug concerning the use of the rand_s function in the Win32 version was fixed.

GNU ccRTP is an implementation of RTP, the real-time transport protocol from the IETF (see RFC 3550, RFC 3551 and RFC 3555). ccRTP is a C++ library based on GNU Common C++ which provides a high performance, flexible and extensible standards-compliant RTP stack with full RTCP support. The design and implementation of ccRTP make it suitable for high capacity servers and gateways as well as personal client applications.
In designing ccRTP, we have taken into account that RTP has been defined as an application level protocol framework rather than a typical Internet transport protocol such as TCP and UDP. Thus, RTP is hardly ever implemented as a layer separated from the application.
Consequently, RTP applications often must customize the adaptable RTP packet layout and processing rules, timing constraints, session membership rules as well as other RTP and RTCP mechanisms. ccRTP aims to provide a framework for the RTP framework, rather than being just an RTP packet manipulation library.
Support for both audio and video data is also considered in the design of ccRTP, that can do partial frame splits/re-assembly. Unicast, multi-unicast and multicast transport models are supported, as well as multiple active synchronization sources, multiple RTP sessions (SSRC spaces), and multiple RTP applications (CNAME spaces). This allows its use for building all forms of Internet standards based audio and visual conferencing systems.
GNU ccRTP is threadsafe and high performance. It uses packet queue lists for the reception and transmission of data packets. Both inter-media and intra-media synchronization is automatically handled within the incoming and outgoing packet queues. GNU ccRTP offers support for RTCP and many other standard and extended features that are needed for both compatible and advanced streaming applications.
It can mix multiple payload types in stream, and hence can be used to impliment RFC 2833 compliant signaling applications as well as other specialized things. GNU ccRTP also offers direct RTP and RTCP packet filtering.
GNU ccRTP uses templates to isolate threading and sockets related dependencies, so that it can be used to impliment realtime streaming with different threading models and underlying transport protocols, not just with IPV4 UDP sockets. For a more detailed list of ccRTP features you can have a look at the programmers manual.
At its highest level, ccRTP provides classes for the real-time transport of data through RTP sessions, as well as the control functions of RTCP.
The main concept in the ccRTP implementation of RTP sessions is the use of packet queues to handle transmission and reception of RTP data packets/application data units. In ccRTP, a data block is transmitted by putting it into the transmission (outgoing packets) queue, and received by getting it from the reception (incoming packets) queue.
Main features:
- Highly extensible to specialized stacks.
- Supports unicast, multi-unicast and multicast. Handles multiple sources (including synchronization sources and contributing sources) and destinations. Also supports symmetric RTP.
- Automatic RTCP functions handling, such as association of synchronization sources from the same participant or NTP-RTP timestamp mapping.
- Genericity as for underlying network and transport protocols through templates.
- It is threadsafe and supports almost any threading model.
- Generic and extensible RTP and RTCP header validity checks.
- Handles source states and information as well as statistics recording.
- Automatically handles SSRC collisions and performs loop detection.
- Implements timer reconsideration and reverse reconsideration.
- Provides good random numbers, based on /dev/urandom or, alternatively, on MD5.
There are several levels of interface (public interface, public or protected inheritance, etc) in ccRTP. For instance, the rtphello demo program distributed with ccRTP just uses the public interface of the RTPSession class and does not redefine the virtual method onGotSR, thus what this program knows about SR reports is the information conveyed in the last sender report from any source, which can be retrieved via the getMRSenderInfo method of the SyncSource class.
On the contrary, the rtplisten demo program redefines onGotSR by means of inheritance and could do specialized processing of these RTCP packets. Generally, both data and control packets are not directly accessible through the most external interface.
All this functions are performed through a few essential classes and types. The most basic ones are the enumerated type StaticPayloadType, and the classes StaticPayloadFormat and DynamicPayloadFormat.
The most important ones are the classes RTPSession, SyncSource, Participant and AppDataUnit, that represent RTP sessions, synchronization sources, participants in an RTP application, and application data units conveyed in RTP data packets, respectively.
When using ccRTP, both sending and receiving of data transported over RTP sessions is done through reception and transmission queues handled by the RTP stack. In the most common case, a separate execution thread for each RTP session handles the queues. This case is the threading model that we will generally assume throughout this document. Note however that ccRTP supports other threading models, particularly ccRTP supports the use of a single execution thread to serve a set of RTP sessions. It is also possible to not associate any separate thread with any RTP session, manually calling the main data and control service methods from whatever other thread.
The basic idea for packet reception with ccRTP is that the application does not directly read packets from sockets but gets them from a reception queue. The stack is responsible for inserting received packets in the reception queue and handling this queue. In general, a packet reception and insertion in the reception queue does not occur at the same time the application gets it from the queue.
Conversely, the basic idea for packet transmission with ccRTP is that packets are not directly written to sockets but inserted in a transmission queue handled by the stack. In general, packet insertion and transmission occur at different times, though it is not necessary.
In order to use ccRTP, you must include the main header (#include < ccrtp/rtp.h >. Two additional headers are provided by ccRTP:
#include < ccrtp/rtppool.h
Classes for pools of RTP service threads.
#include < ccrtp/rtpext.h >
Classes for RTP extensions which are not mature yet.
You must also link in the library, currently ccrtp1.
Enhancements:
- Brand new support has been introduced for Secure RTP Profile (srtp) as per RFC 3711.
- This release also supports a new add-on package, libzrtpcpp, that directly offers native zfone (zrtp) compatible encryption capabilities to Common C++ RTP based applications.
- This is the first softphone client to use both Common C++ RTP srtp and zrtp support.

Sfront project compiles MPEG 4 Structured Audio (MP4-SA) bitstreams into efficient C programs that generate audio when executed. MP4-SA is a standard for normative algorithmic sound, that combines an audio signal processing language (SAOL) with score languages (SASL, and the legacy MIDI File Format).
Under Linux and Mac OS X, sfront supports real-time, low-latency audio input/output, local MIDI input from soundcards, and networked MIDI input using RTP and SIP. A SIP server hosted on the Berkeley campus manages sessions. The documentation includes a book about SAOL programming.
Enhancements:
- This release defaults to writing 16-bit WAV and AIFF files, and the command line flags for specifying 16-bit WAV and AIFF files now work correctly.
- In addition, bugs were fixed in the documentation and implementation of the custom control driver API.

Vat is an audio conferencing application developed by the Network Research Group of Lawrence Berkeley National Laboratory. Source code and pre-compiled binaries are available via anonymous ftp.
The LBNL audio tool, vat, is a real-time, multi-party, multimedia application for audio conferencing over the Internet. Vat is based on the Draft Internet Standard Real-time Transport Protocol (RTP) developed by the IETF Audio/Video Transport working group. RTP is an application-level protocol implemented entirely within vat -- you need no special system enhancements to run RTP. Although vat can be run point-to-point using standard unicast IP addresses, it is primarily intended as a multiparty conferencing application. To make use of the conferencing capabilities, your system must support IP Multicast, and ideally, your network should be connected to the IP Multicast Backbone (MBone).
Vat provides only the audio part of a multimedia conference; video, whiteboard, and session control tools are implemented in separate applications. Our video tool is called vic and our whiteboard tool wb, UCL developed the session directory tool sdr Other related applications include ISIs Multimedia Conference Control, mmcc, the Xerox PARC Network Video tool, nv and the INRIA Video-conferencing System, ivs.
Enhancements:
- Added code to Windows audio driver to convert to 8-bit linear if driver doesnt support 16-bit linear (the code in Win95 thats supposed to do this is apparently broken). Problem reported by dozens of SoundBlaster users.
- Unicast sessions wouldnt work under Windows. (Re-)Incoporated fix from John Brezak that somehow got left out of b1 release.
- Restored vat 3.4 "[net localport]" command (needed by ISI/MIT RSVP additions to vat). Requested by Steve Berson.
- Fixed sitebox display bug on monochrome displays. Reported by Evi Nemeth.
- Fixed bug where SS-10 could get stuck sending a continuous tone if the last byte before a completely silent frame was non-zero. Reported by Steve Deering.
- Correct URL of SCO port on vat homepage. Fix from Shawn McMurdo.
- Incorporated Irix5.3 and 6.x configure and config.h fixes from Andrew Cherenson.
- Incorporated AIX configure and config.h fixes from Christian Zahl.

Oreka is a modular and cross-platform system for recording and retrieval of audio streams.
Oreka project currently supports VoIP SIP and sound device based capture. Recordings metadata can be stored in any mainstream database. Retrieval of captured sessions is web based.
Main features:
Recording and storage
- Record VoIP SIP sessions by passively listening to network packets. Both sides of a conversation are mixed together and each call is logged as a separate audio file.
- Record from a standard sound device (e.g. microphone or line input). Can record multiple channels at the same time. Each recording goes to separate audio files
- Open plugin architecture for audio capture means that the system is potentially capable of recording from any audio source.
- Automatic audio segmentation so that continuous audio sources can be split in separate audio files and easily retrieved later.
- Voice activity detection.
- GSM6.10, A-Law and u-Law compression available in order to save disk space.
- Recording metadata logged to file and/or any mainstream database system.
User interface
- Recordings retrieval can be done using the following criteria (when available):
- Timestamp
- Recording duration
- Direction (for a telephone call)
- Remote Party (for a telephone call)
- Local Party (for a telephone call)
Enhancements:
- A critical bug that could cause Orkaudio to crash given a certain sequence of RTP packets was fixed.
- A SIP detection issue on the Siemens platform (Siemens Optipoint 400) was fixed.

vpnd provides a Virtual Private Network Daemon - encrypted TCP/IP.
vpnd is a daemon which connects two networks on network level either via TCP/IP or a (virtual) leased line attached to a serial interface.
All data transfered between the two networks are encrypted using the unpatented free Blowfish encryption algorithm with a key length of up to 576 bits (may be downgraded to a minimum of 0 bits to suit any legal restrictions).
vpnd is not intended as a replacement of existing secured communications software like ssh or tunneling facilities of the operating system.
It is, however, intended as a means of securing transparent network interconnection across potentially insecure channels.
vpnd acquires a pseudo terminal (a pty/tty device pair) and attaches a SLIP line discipline to it. The effect of this is that vpnd now has its own network interface, a SLIP interface which is named slx where x is some number.
All IP packets sent to this interface are read as a datastream by vpnd and the datastream written by vpnd reappears as IP packets on this interface.
vpnd now encrypts the datastream read and sends it through a TCP connection or over a serial line to its peer vpnd. The datastream received by vpnd from its peer is decrypted and then written to the pseudo terminal.
As vpnd doesnt parse the datastream from the pseudo terminal all packets written by the kernel to the SLIP interface get transported.
Thus vpnd tunnels network traffic between two systems even as it is a user level daemon.
Enhancements:
- fixed minor bug in generic whitening code
- fixed ppp mru setup on Linux
- port to x86_64
- added packetize option for slip/ppp interoperability and rtp header compression (SIP VoIP)
- added smallrtp option for forced use of simple checksum for rtp (SIP VoIP) packets in packetize mode for reduced bandwidth requirements