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Sip Tools

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  No analysis available

Sip Tools is a composite project including several toolkits to enhance JAIN-SIP, Java Media Framework, and similar tools centering on SIP and RTP Media. Iced Java is a Java implementation of RFC 5245 ICE, RFC 5389 STUN and RFC 5766 TURN. The goal of this project is to be as all-encompassing of ... [More] use cases as possible, while imposing a minimal burden on the users of the library to modify their code. RTP Streaming is the most obvious use case, though any P2P Datagram based service is a good candidate for using Iced Java to reduce the programming burden imposed by NATs. [Less]

0 lines of code

0 current contributors

0 since last commit

2 users on Open Hub

Activity Not Available
5.0
 
0 Reviews
I Use This
Mostly written in language not available
Licenses: lgpl3
Tags communications firewall firewallbypass firewalls ice java nat network p2p p2pnetworks peer-2-peer peer2peer 10 more...

erlrtpproxy

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  Analyzed 1 day ago

RTP/RTCP proxy for VoIP and IM applications.

8.26K lines of code

0 current contributors

over 10 years since last commit

2 users on Open Hub

Inactive
4.0
   
0 Reviews
I Use This
Mostly written in Erlang
Licenses: gpl3_or_l...
Tags audio erlang kamailio music_on_hold nat openser opensips recording rtcp rtp ser telecommunications 5 more...

PAMI

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  Analyzed about 4 hours ago

PHP Asterisk Manager Interface ( AMI ) supports synchronous command ( action )/ responses and asynchronous events using the pattern observer-listener. Supports commands with responses with multiple events. Very suitable for development of operator consoles and / or asterisk / channels / peers monitoring through SOA, etc

12.8K lines of code

0 current contributors

over 4 years since last commit

2 users on Open Hub

Inactive
0.0
 
0 Reviews
I Use This
Mostly written in PHP
Licenses: apache_2
Tags ami asterisk call event interface manager object oop oriented php sms telephony 3 more...

rtpengine (former mediaproxy-ng)

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  Analyzed 3 days ago

Kernel-based media relay for VoIP servers.

133K lines of code

18 current contributors

6 days since last commit

2 users on Open Hub

High Activity
5.0
 
0 Reviews
I Use This
Mostly written in C
Licenses: gpl3
Tags audio kamailio nat ngcp proxy rtcp rtp sdp tcp telephony udp video 1 more...

reSIProcate

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  Analyzed 3 days ago

The reSIProcate components, particularly the SIP stack, are in use in both commercial and open-source products. The project is dedicated to maintaining a complete, correct, and commercially usable implementation of SIP and a few related protocols. ReSIProcate is ideally suited to individuals or ... [More] companies that are implementing one of the following SIP applications: * Phones (for example, embedded) * Softphones (any platform) * Gateways * Proxies * B2BUAs * Instant Messaging/Presence Servers or Clients [Less]

1.53M lines of code

11 current contributors

6 days since last commit

2 users on Open Hub

Moderate Activity
5.0
 
0 Reviews
I Use This
Mostly written in C
Licenses: BSD-4-Clause
Tags b2bua c++ dtls library nat pbx presence proxy rtcp rtp sip softphone 6 more...

sipsak

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  Analyzed 24 days ago

A small command line tool for developers and administrators of Session Initiation Protocol (SIP) applications. It can be used for some simple tests on SIP applications and devices.

6.5K lines of code

1 current contributors

over 1 year since last commit

2 users on Open Hub

Very Low Activity
0.0
 
0 Reviews
I Use This
Mostly written in C
Licenses: gpl
Tags sip testing tools voip

baresip

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  Analyzed 2 days ago

Baresip is a portable and modular SIP User-Agent with audio and video support. Features: Audio codecs: AMR, BV32, G.711, G.722, G.722.1, G.726, GSM, iLBC, iSAC, L16, OPUS, Silk, Speex Video codecs: H.263, H.264, H.265, MPEG4, VP8 Audio drivers: Alsa, Coreaudio, Gstreamer, OpenSLES, OSS ... [More] , Portaudio, Windows wave Video sources: FFmpeg avformat, MacOSX qtcapture, MacOSX quicktime, Video4Linux and Video4Linux2, X11 Grabber Video output modules: OpenGL, SDL/SDL2, X11, DirectFB NAT Traversal modules: STUN, TURN, ICE, NAT-PMP Media encryption modules: SRTP, DTLS-SRTP, ZRTP [Less]

100K lines of code

18 current contributors

5 days since last commit

2 users on Open Hub

Moderate Activity
5.0
 
0 Reviews
I Use This
Mostly written in C
Licenses: BSD-3-Clause
Tags android codec library presence rtp sdp sip srtp videoconference voip webrtc

xenagi

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  Analyzed 1 day ago

Xen-Agi is an open source java library that allow you to build java applications that interact with the Asterisk Server for VOIP and PBX functionality. This project supports the FastAGI protocol exclusively and is inspired by the Asterisk-Java project.

3.14K lines of code

0 current contributors

about 16 years since last commit

1 users on Open Hub

Inactive
5.0
 
0 Reviews
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Mostly written in Java
Licenses: apache_2
Tags asterisk fastagi java voip xentelco

pjsip4net

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  Analyzed 1 day ago

A high-level SIP phone API for .Net powered by pjsip project

19.9K lines of code

1 current contributors

8 months since last commit

1 users on Open Hub

Very Low Activity
0.0
 
0 Reviews
I Use This
Mostly written in C#
Licenses: apache_2
Tags c# c#net pjsip rtp sip softphone voip voip_client

OneTeam Desktop

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  Analyzed about 19 hours ago

OneTeam desktop is a nice, featureful XMPP client. It has unique and advanced features such as visual roster, conversation threading, command line, history browsing, remote debugging, correction, Nigle voice calls, and Jingle Relay Nodes. Released under MPL 1.1 and GPL 2.0, it is based on the ... [More] Mozilla Gecko platform and is available as a Firefox extention, or a standalone client for Mac OS X, Linux, and Windows. [Less]

107K lines of code

0 current contributors

almost 12 years since last commit

1 users on Open Hub

Inactive
0.0
 
0 Reviews
I Use This
Mostly written in C
Licenses: gpl, mozilla_p...
Tags chat client communication gecko groupchat im instant_messaging jabber jingle linux macosx messaging 9 more...