traud / asterisk-opus
Asterisk 13 transcoding module: Opus
☆32Updated 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
Sorting:
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆79Updated 10 years ago
- Asterisk 13 transcoding module: AMR-WB☆37Updated 3 years ago
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆208Updated last week
- SIPREC recording server based on drachtio and rtpengine☆89Updated 10 months ago
- A SIP call processing server that can be controlled via nodejs applications☆276Updated this week
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆35Updated 8 years ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆120Updated 3 years ago
- Barebone Opensource Powered SBC☆108Updated 8 months ago
- REST API for sharing IP addresses sending unwanted SIP traffic☆61Updated last year
- Simple config file of Kamailio as Loadblancer for calls and registrations☆32Updated 3 years ago
- ☆79Updated 2 months ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆169Updated 10 months ago
- Sip Express Media Server☆168Updated 3 weeks ago
- G.729 and G.723.1 codecs for Asterisk☆47Updated 7 months ago
- Asterisk 13 transcoding module: 3GPP EVS☆26Updated 3 years ago
- SIPssert Testing Framework is a tool used for facilitating conformity testing of complex VoIP setups☆59Updated last month
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆52Updated this week
- asterisk json utilities☆43Updated 2 years ago
- VoIP signaling and media test automation☆119Updated 2 weeks ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated last year
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆169Updated last year
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 6 years ago
- Kamailio configuration for SIP front-end proxy☆21Updated 12 years ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- Simple SIP command line Softphone Client☆57Updated 6 years ago
- Voip Open Linear Testing Suite☆42Updated last month
- ☆62Updated 2 years ago
- Federated SIP deployment☆35Updated last year
- Easier web calling by providing a layer of abstraction around SIP.js☆64Updated last year
- HEP Fidelity Proxy☆15Updated last year