paneru-rajan / asterisk-sipml5
A complete guide to install Asterisk and use sipml5 with python server. Developed for Audio call using webrtc js library sipml5 and Asterisk's Pjsip.
☆18Updated 7 years ago
Alternatives and similar repositories for asterisk-sipml5:
Users that are interested in asterisk-sipml5 are comparing it to the libraries listed below
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated last year
- Simple config file of Kamailio as Loadblancer for calls and registrations☆30Updated 3 years ago
- ☆35Updated 3 years ago
- Real time web based Asterisk monitoring with Meteor☆27Updated 2 weeks ago
- ☆34Updated last week
- React SIP user agent☆52Updated 5 years ago
- Asterisk Management Interface (AMI) to Web-socket proxy☆87Updated last year
- ☆24Updated last year
- Asterisk Queues Dashboard with amiws☆52Updated last year
- Kamailio config for public/private proxy with rtpengine☆11Updated last year
- SIPREC recording server based on drachtio and rtpengine☆87Updated 7 months ago
- Audio fingerprinting and recognition module for the Asterisk☆10Updated 6 years ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 6 years ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆164Updated 8 months ago
- Call API is a front-end layer for managing advanced SIP call flows. It listens for WebSocket connections and talks JSON-RPC 2.0 over them…☆53Updated last year
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆40Updated 10 years ago
- ☆61Updated 2 years ago
- SIPp Call Scenario for Performance Test☆16Updated 4 years ago
- Web Admin Interface for Kamailio☆101Updated 2 weeks ago
- SIP Express Media Server, very fast and flexible SIP media server☆61Updated this week
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆116Updated 2 years ago
- OpenSIPS + RTPEngine Recording + Speech Recognition in HEP☆21Updated 8 months ago
- Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS☆13Updated 9 years ago
- Community Call Center Module using ECCP Protocol.☆40Updated 7 months ago
- ☆26Updated 7 years ago
- sample sipp scenarios for testing freeswitch☆30Updated 2 years ago
- Register Asterisk on consul☆28Updated 2 years ago
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆106Updated 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
- Multitenant PBX☆29Updated last year