traud / asterisk-opus
Asterisk 13 transcoding module: Opus
☆33Updated 3 years ago
Alternatives and similar repositories for asterisk-opus:
Users that are interested in asterisk-opus are comparing it to the libraries listed below
- Asterisk 13 transcoding module: AMR-WB☆37Updated 3 years ago
- Asterisk 13 transcoding module: 3GPP EVS☆26Updated 3 years ago
- Barebone Opensource Powered SBC☆108Updated 8 months ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆35Updated 8 years ago
- Federated SIP deployment☆35Updated last year
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- ☆35Updated 3 weeks ago
- Simple config file of Kamailio as Loadblancer for calls and registrations☆30Updated 3 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
- Blox route configuration (opensips script)☆22Updated 4 years ago
- REST API for sharing IP addresses sending unwanted SIP traffic☆61Updated last year
- asterisk json utilities☆43Updated 2 years ago
- G.729 and G.723.1 codecs for Asterisk☆46Updated 6 months ago
- SIP Express Media Server, very fast and flexible SIP media server☆63Updated this week
- Kamailio Command Line Interface Control Tool☆53Updated 3 months ago
- 100% Open-Source Packet Capture Agent for HEP☆172Updated this week
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆79Updated 10 years ago
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆208Updated this week
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆119Updated 3 years ago
- Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library☆50Updated 4 months ago
- FreeSWITCH dialer program for VoIP performance tests☆45Updated 7 years ago
- HEP Fidelity Proxy☆15Updated last year
- This Project will provide the inbound sip using that we can route did call to customer ip or customer Phone number☆24Updated 7 years ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated last year
- Stream Asterisk audio over Websockets☆174Updated 8 months ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 6 years ago
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆168Updated 11 months ago
- SIPssert Testing Framework is a tool used for facilitating conformity testing of complex VoIP setups☆59Updated 3 weeks ago
- SIP performance test tool☆38Updated 2 years ago
- RTCAgent is an eBPF powered HEP Agent for HOMER/HEPIC☆39Updated last week