sipwise / rtpengine

The Sipwise media proxy for Kamailio
GNU General Public License v3.0
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SRTP output wanted, but no crypto suite was negotiated #1816

Closed robdyck closed 2 months ago

robdyck commented 2 months ago

rtpengine version the issue has been seen with

Version: git-master-28e9079e

Used distribution and its version

Fedora 39

Linux kernel version used

6.8.4-200.fc39.x86_64

CPU architecture issue was seen on (see uname -m)

x86_64

Expected behaviour you didn't see

negotiate DTLS

Unexpected behaviour you saw

SRTP output wanted, but no crypto suite was negotiated

Steps to reproduce the problem

Make a call from WEBRTC agent to non WEBRTC or the other way. dtls-fail.txt

Additional program output to the terminal or logs illustrating the issue

No response

Anything else?

WEBRTC phones have stopped working. I can't relate it to anything that I may have changed. I tried updating rtpengine to the latest and it didn't help. I have a recent opensips so I tried an older version and that didn't help either. The UAs at either end are not complaining except the call gets dropped due to no media.

I set the log level to 7 and also ran rtpengine with --debug-srtp. I only see the single error. SRTP output wanted, but no crypto suite was negotiated

I am attaching logs.

BTW I joined the rtpengine google group but it is not accepting my email.

robdyck commented 2 months ago

I didn't intend for this to be a bug report. I was looking for ideas. Anyway the issue is solved. The system where the WebRTC agent lives had its ipv6 privacy feature set from "enabled, prefer temporary IP" to "enabled, prefer public IP". The WebRTC agent thought it's only candidate was on ipv4. It ignored the ipv6 in the answer SDP. Oddly The agents at either end, the SIP proxy and rtpengine never complained. That is until one end said BYE because of no media.