We have a system that uses SIP for messaging and webRTC SRTP/RTP for the media. Looking to see if a library exists before writing custom code for load testing. Thanks.
Is there a telecom elixir library for simulating voice and video calls? My team needs to generate a large number of these for load testing.
If you have access to an asterisk server, you can try this one GitHub - entropealabs/ex_ari: An Elixir Library for interfacing with Asterisk the Open Source Communications Software using ARI
I’ve built SIP based apps with it, and also used it to load test the same systems. It’s my library, so if you have any questions feel free to reach out.
Yes! We use FreeSWITCH though.
Maybe something from the https://www.membraneframework.org/
Good to know about Membrane. We might look into SIPp plus custom code for starters.