Realtime Audio Streaming

[shushantsharan]

Design

Detailed Documentation

https://github.com/voiceip/oreka/wiki/Live-Streaming

[kingster]

Few comments

• Please fix code indentation
• Remove non required code flows that do nothing
• Refactor to make this more like a Plugin that can be optionally loaded. No core features should be affected if the plugin is not loaded.
• Other comments are inline.
• Also share sample responses of the API's

[kingster]

Looks much better now :)

Please look at the comments.

[shushantsharan]

@kingster Latest Structure:
Files added :

• srs_librtmp.cpp, srs_librtmp.h --> RTMP client Library
• LiveStreamFilter.cpp, LiveStreamFilter.h --> Rtp packets flow through this filter and are ingested to rtmp server from here ondemand
• LiveStreamSession.cpp, LiveStreamSession.h --> finds session for a nativeCallId and pushed start/stop event to the session
• LiveStreamServer.cpp, LiveStreamServer.h --> HTTP server
• json.h --> support for json

POP based refactoring key updates :

• All code changes added to orkaudio.cpp are removed now.
• LiveStreamServer is started when OrkInitialize() is invoked for LiveStreamFilter . This helps in eliminating almost all the code changes from the rest of the project

Still pending improve channel zipping logic

[vcidst]

Hey, this seems interesting. Did you folks make any headway into this? I've been looking for ways to connect SIP calls to Dialogflow framework.

[kingster]

Hi @vcidst

These changes are stable right now, and we have been using this feature in our production for quite some time, with the only caveat being that only G711 (PCMA/PCMU) is supported.

You can try out with the builds from this pr, and the wiki for setup instructions. Please open a fresh issue in case you face any problems while running the builds.