My online writings. This time its about OpenSIPS 101
☆28Mar 20, 2015Updated 11 years ago
Alternatives and similar repositories for book-opensips-101
Users that are interested in book-opensips-101 are comparing it to the libraries listed below. We may earn a commission when you buy through links labeled 'Ad' on this page.
Sorting:
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Jun 7, 2018Updated 8 years ago
- Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS☆14Apr 24, 2015Updated 11 years ago
- Kamailio config for public/private proxy with rtpengine☆17Nov 21, 2023Updated 2 years ago
- React SIP user agent☆58Oct 21, 2019Updated 6 years ago
- Multiparty-meeting (mediasoup) SIP gateway using Kurento☆28Apr 4, 2019Updated 7 years ago
- GPU virtual machines on DigitalOcean Gradient AI • AdGet to production fast with high-performance AMD and NVIDIA GPUs you can spin up in seconds. The definition of operational simplicity.
- Fork of the Asterisk VOIP software with support for the Opus codec☆14Sep 11, 2012Updated 14 years ago
- Connect as WebRTC with FreeSWITCH using SIPjs☆11Apr 9, 2019Updated 7 years ago
- SIP outbound proxy based on drachtio and freeswitch that includes siprec client functionality☆21Jul 5, 2024Updated 2 years ago
- 用来面向企业通信的FreeSwitch相关的配置、开发书籍☆28Sep 17, 2019Updated 7 years ago
- This is an Ansible module which connects to the target CUCM servers via SSH in order to execute CLI commands and gather outputs☆12Jul 30, 2021Updated 5 years ago
- Load-balancing SIP proxy for Freeswitch☆29Jun 28, 2016Updated 10 years ago
- My online writings. This time its about Asterisk 101☆53Jan 12, 2016Updated 10 years ago
- Testprogram for libre and librem☆11Oct 20, 2021Updated 4 years ago
- SIPp scenarios I use for testing SIP stuff☆305Jul 16, 2024Updated 2 years ago
- Deploy to Railway using AI coding agents - Free Credits Offer • AdUse Claude Code, Codex, OpenCode, and more. Autonomous software development now has the infrastructure to match with Railway.
- nodejs client for accessing rtpengine via ng protocol☆24Jul 18, 2023Updated 3 years ago
- ☆26Jun 5, 2013Updated 13 years ago
- VoIP ID is a Free and Open Source Software to build a hosted VoIP server or hosted IPPBX service. The project is just started, we hope t…☆25Apr 2, 2020Updated 6 years ago
- RTP Cluster is a front-end for multiple RTPproxies☆42May 10, 2024Updated 2 years ago
- Extract wav from pcap (rtp)☆14Jul 17, 2018Updated 8 years ago
- Continously watches and notifies if new MACs are discovered in the local LAN.☆15Dec 17, 2012Updated 13 years ago
- Bare UI for Push-To-Talk and SIP Call/Messaging☆23Nov 25, 2019Updated 6 years ago
- WebSocket Command Line Tool For SIP And Template Data☆44May 23, 2024Updated 2 years ago
- C++ network framework，currently we support http, https, websocket☆14Sep 25, 2019Updated 7 years ago
- End-to-end encrypted cloud storage - Proton Drive • AdSpecial offer: 40% Off Yearly / 80% Off First Month. Protect your most important files, photos, and documents from prying eyes.
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆46Aug 19, 2014Updated 12 years ago
- IDA Pro plugin to view Exports☆16Sep 21, 2018Updated 8 years ago
- Asterisk AGI script that makes use of Microsoft Azure Cognitive Services for text to speech synthesis☆19Jul 30, 2022Updated 4 years ago
- Tool for native number resolution - Cisco Unified Communications Manager - CURRI☆16Aug 11, 2018Updated 8 years ago
- A WebRTC-SIP gateway for AVM Fritz!Box based on Kamailio and rtpengine.☆25Aug 14, 2026Updated last month
- Send xConfiguraiton & xCommand to Cisco TelePresence Endpoints i.e. DX70 & DX80☆13Jun 30, 2016Updated 10 years ago
- OpenAPI specification for the Telnyx API☆13Updated this week
- Various SIP resources.☆260Apr 11, 2026Updated 5 months ago
- Integra Workflow Designer for Asterisk☆14Jun 7, 2022Updated 4 years ago
- Wordpress hosting with auto-scaling - Free Trial Offer • AdFully Managed hosting for WordPress and WooCommerce businesses that need reliable, auto-scalable performance. Cloudways SafeUpdates now available.
- Make test calls to our PSTN phone numbers and SIP/VOIP addresses☆17Mar 1, 2023Updated 3 years ago
- Web dialer SIP based on JsSIP☆17Mar 24, 2014Updated 12 years ago
- Kamailio in docker container with TLS enabled using Let's Encrypt.☆26Jan 11, 2023Updated 3 years ago
- Simple Click2Call for Asterisk PBX platform☆14Jan 9, 2015Updated 11 years ago
- A real time stream trans to other application or server for FreeSWITCH☆42Feb 8, 2025Updated last year
- The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.☆482Jul 16, 2026Updated 2 months ago
- A Fork from http://git.freeswitch.org/git/freeswitch-contrib/tree/dvarnes/java/esl-client☆243Apr 29, 2022Updated 4 years ago