Ojero / jssip-demosLinks
Repository of code using JsSIP
☆14Updated 12 years ago
Alternatives and similar repositories for jssip-demos
Users that are interested in jssip-demos are comparing it to the libraries listed below
Sorting:
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆81Updated 10 years ago
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆191Updated 2 weeks ago
- The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.☆449Updated last week
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆169Updated 3 months ago
- Example applications using SIP.js☆59Updated 8 years ago
- Minimalistic FreeSWITCH configuration as a start for new projects☆82Updated 8 years ago
- FreeSWITCH dialer program for VoIP performance tests☆46Updated 7 years ago
- Kamailio SIP Proxy with Sipwise patches☆63Updated last month
- Sip Express Media Server☆175Updated 4 months ago
- SIPp scenarios I use for testing SIP stuff☆300Updated last year
- Meta-repository to test interop between latest versions of opensips, kamailio, sippy b2bua and rtpproxy☆42Updated 3 months ago
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆42Updated 11 years ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆128Updated 3 years ago
- TESTING - replicated hourly from Google Code SVN using sync2git☆59Updated 9 years ago
- Asterisk Manager Proxy☆29Updated 4 years ago
- Kamailio Command Line Interface Control Tool☆56Updated 7 months ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆301Updated last year
- 100% Open-Source Packet Capture Agent for HEP☆173Updated 4 months ago
- This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.☆57Updated 6 years ago
- Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network☆354Updated 5 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆68Updated this week
- Call Analytics Solution for Freeswitch, Asterisk, Kamailio and other VoIP Switches☆304Updated 3 years ago
- asterisk json utilities☆43Updated 2 years ago
- VoIP signaling and media test automation☆119Updated 2 months ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆175Updated last month
- ctxSip is a Javascript based SIP client that uses WebRTC and WebSockets to connect to your SIP server. The UI is designed to be launched …☆326Updated last year
- Web Admin Interface for Kamailio☆103Updated 6 months ago
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆176Updated last month
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆176Updated last year
- The original single-server, multi-tenant, switch agnostic UI built by 2600Hz☆104Updated 10 years ago