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Sofia-SIP

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  Analyzed over 1 year ago

Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services. The primary target platform for Sofia-SIP is ... [More] GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP is licensed under the LGPL. Actively maintained by FreeSWITCH (https://freeswitch.org/fisheye/changelog/freeswitch/libs/sofia-sip). [Less]

333K lines of code

0 current contributors

over 7 years since last commit

6 users on Open Hub

Activity Not Available
5.0
 
0 Reviews
I Use This
Mostly written in C
Licenses: lgpl21_or...
Tags freeswitch sip voip

pjsip

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  Analyzed about 16 hours ago

pjsip is a professionally supported open source comprehensive multimedia communication library based on the SIP protocol. It is integrated with a rich media and a NAT traversal library supporting the ICE protocol. It is very portable and has a small footprint for embedded use.

801K lines of code

3 current contributors

4 days since last commit

6 users on Open Hub

Moderate Activity
5.0
 
0 Reviews
I Use This
Mostly written in C
Licenses: gpl, pjsip_Sta...
Tags ice sip softphone voip

YXA

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  Analyzed about 11 hours ago

Yxa is both a transaction stateful SIP stack, and a set of SIP server applications.

57.7K lines of code

0 current contributors

over 12 years since last commit

6 users on Open Hub

Inactive
5.0
 
0 Reviews
I Use This
Mostly written in Erlang
Licenses: BSD-3-Clause
Tags erlang sip

XiVO Solutions

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  Analyzed over 1 year ago

XiVO is a complete IP communications system for businesses, based on Asterisk and licensed under GPLv3. • This powerful and scalable solution provides access to a comprehensive set of telephony, voice mail and call center functions. XiVO is an open system, enabling development of new functions ... [More] to tailor it perfectly to your requirements. • The XiVO solution boasts these features: - web interface for easy administration, supervision and operation,, - provisioning server to facilitate mass deployment of telephone services, - CTI server giving access to additional features via the XiVO Client application. • XiVO's flexible design does not restrict the choice of system architecture. [Less]

585K lines of code

26 current contributors

over 1 year since last commit

6 users on Open Hub

Activity Not Available
5.0
 
0 Reviews
I Use This
Mostly written in Scala
Licenses: gpl3, lgpl3
Tags akka asterisk docker framework ipbx isdn ivr pbx phone python sip softswitch 8 more...

rtpproxy

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  Analyzed 1 day ago

The Sippy RTPproxy is a high-performance software proxy for RTP streams that can work together with SIP Express Router (SER), OpenSER or Sippy B2BUA. Originally created for handling NAT scenarious it can also act as a generic media relay as well as gateway RTP sessions between IPv4 and IPv6 ... [More] networks. RTPproxy was developed by Maxim Sobolev and now is being actively maintained by the Sippy Software, Inc. The RTPproxy supports some advanced features, such as remote control mode, allowing building scalable distributed SIP VoIP networks. The nathelper module included into the SIP Express Router (SER) or OpenSER as well Sippy B2BUA allow using multiple RTPproxy instances running on remote machines for fault-tolerance and load-balancing purposes. [Less]

277K lines of code

4 current contributors

7 days since last commit

6 users on Open Hub

Moderate Activity
5.0
 
0 Reviews
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Mostly written in C
Licenses: BSD-2-Clause
Tags kamailio nat opensips proxy reSIProcate rtcp rtp sdp ser sip sippy tcp 4 more...

QuteCom (formerly WengoPhone)

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  No analysis available

QuteCom is the new name for the open source softphone previously known as WengoPhone, a standards-based softphone and multi-protocol IM client. It is a community project focussed on communication over IP, including VoIP, instant messaging and video phonecalls.

0 lines of code

0 current contributors

0 since last commit

5 users on Open Hub

Activity Not Available
3.83333
   
0 Reviews
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Mostly written in language not available
Licenses: gpl
Tags c c++ client instant_messenger python qt sip softphone video videophone voip xmpp

Adhearsion

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  Analyzed about 11 hours ago

Adhearsion is an open-source voice application development framework written in Ruby. Adhearsion users write applications atop the framework with native Ruby syntax and a simplified Domain-Specific Language for call management enabling users to call into their code. Adhearsion rests above a ... [More] lower-level telephony platform, namely Asterisk, and provides a framework for integrating with various resources, such as SQL, LDAP and XMPP (Jabber). Adhearsion has... *An elegant dialplan system for writing the code which controls a live phone call *A sophisticated Asterisk Manager Interface library *An events subsystem *A reuseable component architecture *Ability to re-use existing Ruby on Rails database models with ActiveRecord/ActiveLDAP *Easy interactive communication via XMPP instant messages [Less]

64.6K lines of code

6 current contributors

over 3 years since last commit

5 users on Open Hub

Inactive
4.2
   
0 Reviews
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Mostly written in Ruby
Licenses: mit
Tags activerecord agi ami asterisk freeswitch jruby ldap mri ruby ruby19 rubygem rubygems 6 more...

Elastix

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  Analyzed about 19 hours ago

Elastix is an appliance software that integrates the best tools available for Asterisk-based PBXs into a easy-to-use interface. It also adds its own set of utilities to make it the best software package available for open source telephony.

430K lines of code

0 current contributors

over 10 years since last commit

4 users on Open Hub

Inactive
4.33333
   
0 Reviews
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Mostly written in PHP
Licenses: AFL1-1, gpl
Tags appliance asterisk distro fax h323 iax ivr jabber linux mail pbx php 8 more...

SEMS

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  Analyzed about 16 hours ago

Sems is a extensible media server which helps you adding voice services to your VoIP system.

290K lines of code

16 current contributors

14 days since last commit

4 users on Open Hub

Moderate Activity
5.0
 
0 Reviews
I Use This
Mostly written in C++
Licenses: gpl
Tags conferencing ivr rtp sip telephony voice voicemail voip

Sippy SIP B2BUA

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  Analyzed about 16 hours ago

Sippy B2BUA is a RFC3261-compliant SIP Back-to-back user agent (B2BUA). Unlike a SIP proxy server, which only maintains transaction state, the B2BUA maintains complete call state and participates in all call requests. For this reason it can perform number of functions that are not possible to ... [More] implement using SIP proxy, such as for example accurate call accounting, pre-paid rating and billing, fail over call routing etc. Unlike PBX-type solutions such as Asterisk for example, the B2BUA doesn't perform any media relaying or processing, therefore it doesn't introduce any additional packet loss, delay or jitter into the media path. Sippy B2BUA supports RADIUS AAA protocol, which is compatible with Cisco VoIP gateways allowing it to be seamlessly integrated into existing networks. [Less]

11.2K lines of code

1 current contributors

23 days since last commit

3 users on Open Hub

Low Activity
0.0
 
0 Reviews
I Use This
Mostly written in Python
Licenses: gpl
Tags b2bua rtp sip sippy softswitch telephony voip