[Freeswitch-users] Something broken in RTP in latest trunk (r13502)?

[Peter Olsson]

After using the latest trunk revisions I get no audio anymore. The last working build I have is about 5 days ago. I havn't upgraded until today, so I don't know exactly when this happened.

I've noticed quite a few changes on the RTP stack, beacuse of the implementation om ZRTP, and I guess it's somewhere around this time when it happened. How to continue debugging on this issue? I have both a working version of FS (compiled 5 days ago), and a broken one (compiled today), so I can test this very easily, and everything is on a non live server.

The conf-dir is the same between the revisions.

The calls I'm trying to do is both directly to FS (voicemail or similar applications), and aslo calls to another SIP-trunk, to PSTN (media stream is sent through FS).

Regards,

Peter

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