We need to develop a pjsip-based app to bridge calls between WebRTC endpoints and Legacy sip-phones, as below:

[WebRTC endpoint on a browser] <======= webrtc media flow =======> [desired pjsip-based app] <======== sip+rtp flow =======> [SIP/RTP legacy sip-phone]

How can pjsip help with that?

Thanks for replying.

What I am looking for is creating a self-contained component of signalling and media (let’s call it SMLIB) that can be used as part of other, higher-level, components.

As one hypothetical example SMLIB could be used to power a softphone that “talks webrtc”,

Or inside a conference server allowing both legacy SIP and WebRTC-browser clients.

Following your suggestion I should say that OpenSIPS is not a webrtc frontend, it is a medialess (deals only with SIP) component, although in more recent versions it brings WebSockets as a new SIP transport (adding to UDP, TCP, TLS).

Asterisk really supports WebRTC but I suppose I would need to rip-off the webrtc parts of its source code base to use in my Lib, that could be challenging.

Any other thoughts?

Thank you for the recommendation.
Downloaded SylkServer source on a Ubuntu server to experiment it.

two questions:

• what is the best place to learn the details of APIs, configuration and code integration?

• would it be easy to integrate SylkServer in with another running process that would control SylkServer and receive ‘callbacks’ from it in realtime ?