Free initiation software for linux

Bio::Tools::Run::PiseApplication::fasta is a Bioperl class for sequence database search.

Parameters:

fasta (Excl)
Fasta program

query (Sequence)
Query sequence File
pipe: seqfile

seqtype (Excl)
Is it a DNA or protein sequence (-n)

protein_db (Excl)
Protein Database

nucleotid_db (Excl)
Nucleotid Database

break_long (Integer)
Break long library sequences into blocks (-N)

ktup (Integer)
ktup : sensitivity and speed of the search (protein:2, DNA:6)

optcut (Integer)
OPTCUT : the threshold for optimization. (-c)

gapinit (Integer)
Penalty for gap initiation (-12 by default for fasta with proteins, -16 for DNA) (-f)

gapext (Integer)
Penalty for gap extention (-2 by default for fasta with proteins, -4 for DNA) (-g)

high_expect (Float)
Maximal expectation value threshold for displaying scores and alignments (-E)

low_expect (Float)
Minimal expectation value threshold for displaying scores and alignments (-F)

nucleotid_match (Integer)
Reward for a nucleotid match (-r)

nucleotid_mismatch (Integer)
Penalty for a nucleotid mismatch (-r)

matrix (Excl)
Scoring matrix file (-s)

X_penalty (Integer)
Penalty for a match to X (independently of the PAM matrix) (-x)

frameshift (Integer)
Penalty for frameshift between codon (fast[xy]/tfast[xy])(-h)

frameshift_within (Integer)
Penalty for frameshift within a codon (fasty/tfasty)(-j)

threeframe (Switch)
Search only the three forward frames (tfasta) (-3)

invert (Switch)
Reverse complement the query sequence (all tfasta) (-i)

genetic_code (Excl)
Use genetic code for translation (tfasta/tfast[xy]/fast[xy]) (-t)

band (Integer)
band-width used for optimization (-y)

swalig (Switch)
unlimited Smith-Waterman alignment for DNA (-A)

noopt (Switch)
no limited optimization (-o)

stat (Excl)
Specify statistical calculation. (-z)

random (Switch)
Estimate stat parameters from shuffled copies of each library sequence (-z)

histogram (Switch)
No histogram (-H)

scores (Integer)
number of similarity scores to be shown (-b)

alns (Integer)
number of alignments to be shown (-d)

html_output (Switch)
HTML output (-m)

markx (Excl)
Alternate display of matches and mismatches in alignments

init1 (Switch)
sequences ranked by the z-score based on the init1 score (-1)

z_score_out (Excl)
Show normalize score as (-B)

showall (Switch)
both sequences are shown in their entirety in alignments (fasta only) (-a)

linlen (Integer)
output line length for sequence alignments (max. 200) (-w)

offsets (String)
Start numbering the aligned sequences at position x1 x2 (2 numbers) (-X)

info (Switch)
Display more information about the library sequence in the alignment (-L)

statfile (OutFile)
Write out the sequence identifier, superfamily number, and similarity scores to this file (-R)

filter (Switch)
Lower case filtering (-S)

outfile (OutFile)
pipe: mview_input

html_outfile (OutFile)

The Wengophone is an easy to use program, that enables you to call anyone, anywhere, anytime, for free. Its native use of the SIP protocol makes it interoperable with most on the known VoIP technologies.
WengoPhone already has built-in video capabilities, and is will offer many more features in the near future such as calling ordinary phones (land lines and mobiles) at very low prices, conference calls, user communities and innovative cellphone-based services.
With our software, you have better audio quality, broader connectivity and low bandwidth usage. Additionally, thanks to its GPL licence and SIP-compliant foundations, you get the freedom of open source software: add functionnality or modify Wengophone to fit your needs.
We are aware of the challenge that is launching an open source project, not just in terms of creating a new technology, but also in getting developers and future customers to support the efforts. We are commited to consistent and honest communication.
OpenWengo and Wengos commercial ventures are complementary, and our goal is to provide an alternative to expensive and/or proprietary services.
We expect to compete strongly with Skype in the realm of software-based services (PC to PC) and complementary access modes (ATA, mobile devices). The trick? Free calls, Free software.
SIP, short for Session Initiation Protocol, is an open standard developed by the Internet Engineering Task Force. Eventually, it will enable interoperability with other compliant VoIP service providers. One will be able to call any other SIP-compliant softphone user, seamlessly and for free. Moreover, SIP will enable new services for mobile users.
Main features:
- MakingCalls
- ChatSupport
- VideoSupport
- Presence
- EmotiSounds
- FileTransfers
Wengo uses your broadband internet connection (DSL, Cable or WiFi) to provide telephony services. Call your friends, family, and colleagues anywhere, anytime, for free. Use video and chat features, also for free.
Wengo also offers low cost international rates (prepaid or unlimited) to land lines or mobile phones, and an SMS gateway service.
Wengo provides its customers with a Wenbox ATA, that enables the use of any regular phone. This device converts analog phone signals so that calls can be placed over the internet.

sipsak is a small comand line tool for developers and administrators of Session Initiation Protocol (SIP) applications. sipsak can be used for some simple tests on SIP applications and devices.
Main features:
- sending OPTIONS request
- sending text files (which should contain SIP requests)
- traceroute (see section 11 in RFC3261)
- user location test
- flooding test
- random character trashed test
- interpret and react on response
- authentication with qop supported
- short notation supported for receiving (not for sending)
- string replacement in files
- can simulate calls in usrloc mode
- uses symmetric signaling and thus should work behind NAT
- can upload any given contact to a registrar
- send messages to any SIP destination
- Nagios compliant return codes
- search for strings in reply with regluar expression
- use multiple processes to create more server load
- read SIP message from STDIN (e.g. from a pipe |)
- supports DNS SRV through libruli
Version restrictions:
- The hostname is used in the Via line, which is not correct in all cases (e.g. if the loopback interface is used, or if the host has several interfaces). The rport parameter should fix problmes with the incorrect hostname, but for backward compatibility whith implementations which do not support rport this should be fixed.
- The DNS responses are not parsed compeltly which can result in strange output during hostname detection.
- TCP is not supported as transport protocol.
- IPv6 is not supported as transport protocol.
- Missing support for the Record-Route and Route header.
- Not fully RFC3261 compatible.
- Some smaller problems are listed in the TODO file.
Enhancements:
- A new option allows to add any header to the outgoing requests.
- The variable replacement option now accepts any number of attribute value pairs.
- Besides MD5 now SHA1 is support as digest authentication algorithm.
- The password for authentication can be read from stdin to prevent password disclosure in the process list.
- Fixed problems when executed as user root and compiles fine again under cygwin.

Siproxd is a proxy/masquerading daemon for the SIP protocol. It handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections work via an masquerading firewall (NAT).
Siproxd project allows SIP software clients (like kphone, linphone) or SIP hardware clients (Voice over IP phones which are SIP-compatible, such as those from Cisco, Grandstream or Snom) to work behind an IP masquerading firewall or NAT router.
SIP (Session Initiation Protocol, RFC3261) is the protocol of choice for most VoIP (Voice over IP) phones to initiate communication. By itself, SIP does not work via masquerading firewalls as the transfered data contains IP addresses and port numbers.
There do exist other solutions to traverse NAT existing (like STUN, or SIP aware NAT routers), but such a solutions has its disadvantages or may not be applied to a given situation. Siproxd does not aim to be a replacement for these solutions, however in some situations siproxd may bring advantages.
HOW TO GET STARTED
make sure libosip2 is installed
If your libposip2 libraries are installed in /usr/local/lib, be sure to include this library path to /etc/ld.so.conf
$ ./configure
$ make
$ make install
edit /usr/etc/siproxd.conf according to your situation.
At least configure if_inbound and if_outbound. They must represent the interface names (e.g. on Linux: ppp0, eth1) for the inbound and outbound interface.
edit /usr/etc/siproxd_passwd.cfg if you enable client authentication in siproxd.conf
start siproxd (siproxd does not require root privileges)
$ siproxd
Enhancements:
- Several issues related to 64 bit architectures have been fixed and several minor bugfixes.

RtspProxy is a proxy server for multimedia streaming services based on the RTSP protocol.
The current version is a complete rewrite from scratch in Java of previous versions based on C++. The goal is to build a robust and scalable system usable in production environment.
The proxy is based on an asynchronous network framework, Apache MINA , which is built on Java NIO. This framework does permit to RtspProxy to handle high loads and concurrent users.
RTSP Caching Proxy is a proxy server that works with the RTSP protocol in multimedia streaming reproduction. RTSP (Real Time Streaming Protocol) is an emerging protocol for "session initiation" scopes, its purpose is to establish the conditions of an audio-video streaming session. The streaming data is then sent over other channels using other protocol such as RTP (Real Time Protocol).
Using a RTSP proxy does permits to access multimedia content without the need of being directly connected to the internet, for example being behind a corporate firewall.
Enhancements:
- A preliminary implementation of the IP address and host name filter was added.
- The Windows startup script was fixed.
- Session data is properly cleared when a session is closed.
- Only a UDP port pair is used for all RTP/RTCP packet handling and sending for all the connected clients.

sipX is a next generation IP PBX solution offering rich functionality combined with ease of use, installation, and administration. sipX is entirely based on the Session Initiation Protocol (SIP) and significant attention is paid to standards compliance and interoperability.
It combines all common calling features, XML-based SIP call routing, Web-based configuration, and integrated management and configuration of the PBX and attached phones and gateways.
It is a modular server-based solution that does not require any additional hardware, as it interoperates with any SIP compliant gateway, phone, or application.
Main features:
- Native SIP IP PBX Solution
- Voicemail
- Multiple auto-attendants
- Presence
- SIP call routing
- Web based management
- Scalable architecture

ccgfs is a transport-agnostic FUSE filesystem. Transport is arranged by helper programs, such as ssh. The "push" transport mode makes it possible to export a filesystem located in a LAN to a DMZ host without defeating the DMZ security model which prohibits connection initiation to the LAN.