On-Hands Dev Session: Fighting the RTP Bleed (OpenSIPS & RTPProxy)

On-Hands Dev Session: Fighting the RTP Bleed at SIP / RTP level

We’re getting together for a hands-on hacking and discussion session around the long-standing RTP Bleed issue that recently got to a spotlight (again) the SIP / RTP / WebRTC world.

Sandro Gauci (Enable Security) — the guy who first dug into it — will join us to explain what’s going on and what we can do about it.

Also joining are core devs from OpenSIPS and RTPProxy, so this should be a great mix of people who actually build and run this stuff.

We’ll go over:

• What RTP Bleed actually is and how it works in the wild
• How it affects SIP setups and RTPProxy deployments
• Current fixes, patches, and config tweaks
• Ideas for longer-term hardening on both signaling and media levels

Expect a relaxed but technical session — part talk, part live problem-solving, maybe a few rants about NAT and “learning” modes.

If you run SIP infrastructure or just like poking at media paths, you’ll fit right in.

Time permitting, we might also look at the question of packaging third-party OpenSIPS modules (such as rtp.io, or any other with complex dependency chain).