v1.7.0 - Agents

Website for better Documentation

• Github page for better docs -> emiago.github.io/diagox

New Feature: Diagox Agents 🚀

• Experimental - Agents Framework as RPC feature for those who like to route and control calls in programmable way over simple Request/Response API. More on -> emiago.github.io/diagox/docs/agent_rpc/)
• NodeJS/Typescript SDK provided to make easier way of usage. Checkout some of Examples [github.com/emiago/diagox-agent-sdk-js/tree/main/examples](https://github.com/emiago/diagox-agent-sdk-js/tree/main/examples
• Protocol is custom built as WS/JSON but very simplified for now. Of course I am aware some "better" alternatives, but goal more is easy integration and debugging.
• Allows connecting multiple agents for redundancy or load balancing

NOTE: Features is experimental. IN case you use it, DO NOT expose RPC Port on public network!

Other Highlights

• Update to latest diago lib and sipgo
• Docker image as emiago/diagox for more easier running in containerized env.
• GUI with Call History improved and lot of fixes. FRONTEND_ENABLE=true needed

Planned but no deadlines

• More Agent RPC SDK work support.
• Outgoing call Agent RPC
• Webrtc phone in GUI for testing

v1.6.0

Highlights

Adding support for overriding contact header for outgoing INVITE

endpoints:
  bob: # Address of record
    uri: "sip:localhost:5080" 
    contact_header: "<sip:override@127.0.0.1:5060;id=1;+sip_instance=123>" # Override contact header with custom one

v1.5.0

WebRTC secure 🔒

Webrtc was released in previous version, but to make use of it, normally secure websocket is needed.
This release adds docs and extends behavior

TLS Server configuration

Supporting TLS configuration either base64 encoded (easier for cloud k8s) or via path.

// SIP SERVER TLS configuration
string "SERVER_TLS_KEY" envDefault:"" // Base64 encoded key
string "SERVER_TLS_CRT" envDefault:"" // Base64 encoded crt
string "SERVER_TLS_KEY_PATH" envDefault:""
string "SERVER_TLS_CRT_PATH" envDefault:""
//

Call History Legs GUI sneak peak

Call history exposed in GUI is refactored.
NOTE: GUI is work in progress and not consider stable

v1.4.0 SIP <-> WEBRTC

SIP - WEBRTC RELEASE 🚀 ☎️ 🌎

This release brings many fixes, but major feature is now WEBRTC OVER SIP bridging
It allows bridging your Browser phones with Standard SIP telephony and vice versa.

Features:

• 📞 SIP WEBRTC REGISTRATION for accepting Calls support
• 📞 Accepting WEBRTC calls and bridging to SIP telephony or other WEBRTC media capable endpoint

This makes now diagox full compatible with today Browser Softphone needs. :chro

Configuration

Example of configuration of endpoint

endpoints:
 webrtc:
    match: 
      type: "user" 
    auth:
      username: "webrtc" 
      password: "webrtc" 
    media:
      type: "webrtc" 

Extra

Diagox already supports built IN Webrtc softphone 🚀 ☎️ , but this will not be exposed for now.

v1.2.0

• Update to latest diago v0.16.0 release
• Added build info flag
• Fixed minority issues

v1.1.0

Call history in memory improvements

v1.0.0

Initial of diagox release.