Realtime Audio Streaming

shushantsharan commented 3 years ago

Design

Detailed Documentation

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

shushantsharan commented 3 years ago

@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

vcidst commented 2 years ago

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 commented 2 years ago

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.

voiceip / oreka

Realtime Audio Streaming #45