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Restcomm

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Claimed by TeleStax Analyzed 4 months ago

Restcomm is the only full Stack Communications Platform as a Service (cPaaS). It enables you to create, deploy and manage services and applications integrating voice, video and data across a range of IP and legacy communications networks. It drives convergence with the following key enablers ... [More] : Communications API Layer and Visual Designer in Restcomm Connect, WebRTC SDKs for Web and Mobile, SMSC, USSD Gateway, GMLC for GeoLocation. It also offers middleware telecom infrastructure with JAIN-SLEE, SIP Servlets, SS7 Stack, SIP Stack, Diameter Stack and SMPP Stack Restcomm is supported by TeleStax [Less]

5.54M lines of code

20 current contributors

about 2 years since last commit

10 users on Open Hub

Activity Not Available
4.375
   
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rtpproxy

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  Analyzed about 19 hours 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]

113K lines of code

4 current contributors

about 2 months since last commit

6 users on Open Hub

Very Low Activity
5.0
 
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baresip

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  Analyzed about 14 hours 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

1 day since last commit

2 users on Open Hub

Moderate Activity
5.0
 
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rtpengine (former mediaproxy-ng)

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  Analyzed about 3 hours ago

Kernel-based media relay for VoIP servers.

123K lines of code

18 current contributors

2 days since last commit

2 users on Open Hub

High Activity
5.0
 
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sngrep

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  Analyzed about 23 hours ago

SIP callflow viewer using ngrep

15K lines of code

4 current contributors

11 days since last commit

1 users on Open Hub

Very Low Activity
5.0
 
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Callflow Sequence Diagram Generator

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  Analyzed 4 months ago

The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by wireshark and produce a time sequence diagram. This is useful to view & debug SIP callflows or other network traffic

5.72K lines of code

0 current contributors

about 7 years since last commit

1 users on Open Hub

Activity Not Available
0.0
 
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SDPAda

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  No analysis available

The objective of this project is the construction of a package Ada for parsing and building session descriptions in SDP format (RFC 4566)

0 lines of code

0 current contributors

0 since last commit

0 users on Open Hub

Activity Not Available
0.0
 
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Mostly written in language not available
Licenses: mit

creytiv-libre

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  Analyzed about 19 hours ago

Libre is a portable and generic library for real-time communications with async IO support and a complete SIP stack with support for SDP, RTP/RTCP, STUN/TURN/ICE, BFCP and DNS Client. Features SIP Stack (RFC 3261) SDP RTP and RTCP SRTP DNS-Client STUN/TURN/ICE BFCP HTTP-stack with ... [More] client/server Websockets Jitter-buffer Async I/O (poll, epoll, select) UDP/TCP/TLS/DTLS transport [Less]

39K lines of code

0 current contributors

over 8 years since last commit

0 users on Open Hub

Inactive
5.0
 
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