traud / asterisk-amr
Asterisk 13 transcoding module: AMR-WB
☆37Updated 3 years ago
Alternatives and similar repositories for asterisk-amr:
Users that are interested in asterisk-amr are comparing it to the libraries listed below
- Asterisk 13 transcoding module: 3GPP EVS☆26Updated 2 years ago
- This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.☆54Updated 6 years ago
- Barebone Opensource Powered SBC☆108Updated 6 months ago
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆108Updated 2 years ago
- Kamailio Command Line Interface Control Tool☆51Updated 2 months ago
- SIP performance test tool☆38Updated 2 years ago
- ☆34Updated this week
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆169Updated 9 months ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 6 years ago
- A library to replace the monolithic Kemi example file, making it testable☆16Updated last year
- VoIP signaling and media test automation☆118Updated last week
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆117Updated 3 years ago
- Black Box SIP Tester☆31Updated 3 months ago
- Voip Open Linear Testing Suite☆42Updated last month
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆43Updated 3 years ago
- ☆16Updated 8 years ago
- Sip Express Media Server☆165Updated 3 months ago
- G.729 and G.723.1 codecs for Asterisk☆45Updated 5 months ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated last year
- Kamailio+rtpengine failover demo infrastructure for kamailio-ru-2020 conference☆15Updated 3 years ago
- 100% Open-Source Packet Capture Agent for HEP☆172Updated last week
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆178Updated last week
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆35Updated 8 years ago
- Federated SIP deployment☆35Updated last year
- FreeSWITCH dialer program for VoIP performance tests☆44Updated 7 years ago
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆171Updated 3 years ago
- SIPREC recording server based on drachtio and rtpengine☆87Updated 8 months ago
- SIPssert Testing Framework is a tool used for facilitating conformity testing of complex VoIP setups☆56Updated last month
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆204Updated this week
- Simple config file of Kamailio as Loadblancer for calls and registrations☆30Updated 3 years ago